Quantcast
Channel: General Help - FreePBX Community Forums
Viewing all 12616 articles
Browse latest View live
↧

Custom dialplan is not recording any calls

June 20, 2018, 11:22 am
$
0
0

@faisalkhan wrote:

Hi all,

I need a little help here with my dialplan.

I am passing some dtmf in my code and those calls are not recorded. I want them to be recorded as well.

example of code shown below:

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;; ;;
;; Entering Dialplan Direct Dialing ;;
;; ;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

exten => 221,1,Set(CALLERID(num)=123456789)
exten => 221,n,NoOp(I am in Quick Dialing)
exten => 221,n,Dial(SIP/SIPROUTES/987654321,20,M(221))

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;; ;;
;; Entering Macro of Direct Dialing ;;
;; ;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[macro-221]
exten => s,1,Answer()
exten => s,n,Wait(1)
exten => s,n,SendDTMF(221)
exten => s,n,Wait(1)

I want to record these dialcodes. when someone dial these dialcodes they will be recorded as the other calls are recorded.

And also one question can I use outbound route in place of my carrier (SIPROUTES) in this dialplan?

Posts: 7

Participants: 2

Read full topic

↧
Search

Twilio vs Les.Net

June 20, 2018, 1:27 pm
$
0
0

@eagle wrote:

I’ve been using Les.Net for many years as I found it to be a reasonably priced SIP provider with decent customer service.

I was shopping around a couple of months ago and realized that now they are pretty expensive compared to others so I started checking out other SIP providers, specifically any providers that would also have good API’s and/or integration into my other systems (like my CRM) and found Twilio. Their plans are about 1/2 that of Les.Net and I’ve been running a test account with them tied into my PBX now for a couple of months and it seems to be going well. All of our outbound calls have been on Twilio and we’ve been pretty happy with their SIP service.

I’m curious what others think between these two providers or if they think Twilio is not a good SIP provider.

Posts: 1

Participants: 1

Read full topic

↧
↧

TLS error (everything else working)

June 21, 2018, 3:38 am
$
0
0

@clopezi wrote:

Hi,

I have a fresh installation of FreePBX with the official distro. I have all working fine, but TLS.

I have enabled TLS on SIP configuration, and also in the extension, who says i can connect for 5160 for UDP and 5161 for TLS.

If i connect trough UDP, the soft connects (zoipper, 3cx…) and the asterisk log is normal:

[2018-06-21 10:14:30] VERBOSE[9761] chan_sip.c: Registered SIP '4' at X.X.X.X:50482
[2018-06-21 10:14:30] NOTICE[9761] chan_sip.c: Peer '4' is now Reachable. (167ms / 2000ms)

But if i connect trought TLS, the soft doesn’t connect and the asterisk log says:

[2018-06-21 10:13:35] ERROR[11876] tcptls.c: Problem setting up ssl connection: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[2018-06-21 10:13:35] WARNING[11876] tcptls.c: FILE * open failed!

I have tried with Let’s Encrypts SSL (working on the domain) and self-signed, but both of them shows the same error.

Any cloues?

King regards and thank you!

Posts: 1

Participants: 1

Read full topic

↧

SIPTAPI, Windows 10, Outlook 2016

June 21, 2018, 4:44 am
$
0
0

@vezonetworks wrote:

Hi PBX gurus!

been asked to get click-to-dial working through outlook on our in-house FreePBX and having a little trouble.

I’ve pub SIPTAPI onto my machine as a test by copying the relevant .tsp file into my system32 folder. Entered my extension details into it and seems to be fine. however!

When I click on my directors contact (internal extension) it dials my desk phone as expected, but when I pick up it does not then dial the destination. Any thoughts or alternatives?

Posts: 1

Participants: 1

Read full topic

↧

Time Conditions Control Manually for Perminant Settings

June 21, 2018, 7:18 am
$
0
0

@jessy5765 wrote:

We have a few clients that would like to be able to set Permanently Matched/Unmatched for the time conditions from their phones for instance if they decided they are not coming in for the morning that it never goes back to the normal condition and set it Open business hours.

Posts: 1

Participants: 1

Read full topic

↧
↧

Retrieve_conf failed after upgrade script 10.13.66-22

June 21, 2018, 8:18 am
$
0
0

@cassius wrote:

I just ran upgrade-10.13.66-22.sh, and retrieve_conf fails with 255. I got the following at the system prompt:

[root@vieasterisk2 ~]# fwconsole reload
Reloading FreePBX
Error(s) have occured, the following is the retrieve_conf output:
exit: 255
Unable to continue. died in splice ext-did-0002 s in /var/www/html/admin/libraries/extensions.class.php on line 197
#0 /var/www/html/admin/modules/blacklist/functions.inc.php(58): extensions->splice(‘ext-did-0002’, ‘s’, ‘did’, Object(ext_set))
#1 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(95): blacklist_hookGet_config(‘asterisk’)
#2 /var/lib/asterisk/bin/retrieve_conf(864): FreePBX\DialplanHooks->processHooks(‘asterisk’, Array)
#3 {main}

Posts: 2

Participants: 2

Read full topic

↧

IAX external phones dropping and not reconnecting

June 21, 2018, 8:47 am
$
0
0

@gregorywest wrote:

I have a few IAX softphones running on Android. The problem I am having is when the phone moves from one hotspot to another (on WiFi) the registration of the phone drops and does not get reconnected for a long time. In that time incoming calls to the smartphone are dropped (sent to voicemail) instead of causing the phone to ring.

If there a down and dirty way to fix this?

Greg

Posts: 1

Participants: 1

Read full topic

↧

Paging not working

June 21, 2018, 8:58 am
$
0
0

@gregorywest wrote:

I have a mix of Aastra 6755i / 6757i and PolyCom 501’s. I have accepted I will never get paging to work on the Polycoms. For some reason I can not get paging to work to the Aastra’s either. I dial *80ext# and all that happens is I hear one ring, then Asterisk play the busy signal.

Do I have something configured wrong? The Aastra phones are set up not to auto answer, This might be a reason, but I do not want the phones to automatically go to speaker when someone calls. With the exception of pages.

Posts: 2

Participants: 2

Read full topic

↧
Search

How to remove old CDRs

June 21, 2018, 11:38 am
$
0
0

@bksales wrote:

Ive got a couple of PBXs that have millions upon millions of lines of CDRs going back 5-10 years and its getting impossible to run backups or restores. Is there an easy way to export and delete old CDRs, say anything older than a year?

I read the article in the wiki about remote cdr databases but there isn’t much info there about how to remove old CDR lines, just how to start recording everything on the new remote server. Or will it move the entire CDR database to the remote location once I set that up? Not sure how that would effect QXact reporting.

Or is there an easy way to just backup a certain date range of CDR? I would consider saving a CDR backup for each year somewhere remote.

Thanks.

https://wiki.freepbx.org/display/FPG/Remote+CDR

Posts: 2

Participants: 1

Read full topic

↧
↧

Asterisk Click2Dial Firefox Addon with FreePBX

June 21, 2018, 5:00 pm
$
0
0

@stevensedory wrote:

Hey all,

On FreePBX 14.0.3.2 with Asterisk 15.1.4

Trying to get Asterisk Click2Dial 0.3 working, but can’t. It needs AJAM.

Have manager.conf including an additional like webenable=yes. Also have enable the mini-HTTP server under Advanced Settings, but no go.

Just doesn’t work. Not sure best way to troubleshoot.

Here’s a screenshot of the settings:

image

And this seems to show it is working, when you go to: http://23.X.X.X:8088/httpstatus

image

Posts: 1

Participants: 1

Read full topic

↧

IP Phone Extentions NOT registering with PBX

June 21, 2018, 5:02 pm
$
0
0

@a01020304 wrote:

I am having trouble getting the extensions to work with my phones.
I have set up Trunks, Incoming and Outgoing Routes, Extenstions and Ring Groups as per install instructions.
I added the extensions to my Polycom Soundpoint IP601 phone and the extensions show on display hopwever the extensions are not connecting with the pbx.

When I do a test call to my phone number the phone rings normally and can answer it and speak to people. the call routs directly to my phone and does not seem to go through pbx.
When I make an outgoing call can also speak to people so acts normally.
If I dial just the extension number on its own it connects directly to voip provider and get recoding says cannot connect to this number, so the extention dialed is routing directly to voip provider and not going through pbx.

so it seems the incoming and outgoing routes are set up fine but its the extentions issues that are baffling me

i have taken screenshots of pbx and phone settings so can anyone tell me what is wrong?

click on image below to see all 19 screenshot images

[Album] Imgur

please note: this is a test account to get to know system so passwords and other stuff is simple for a reason and not used on LIVE system.

Posts: 1

Participants: 1

Read full topic

↧

Cant Make Call using callcentric

June 21, 2018, 5:47 pm
$
0
0

@kjones9999 wrote:

Hi all. 8 hrs is enough trying.

I have a call centric account that registers. I am using the latest Freepbx install.

I cant make outbound calls despite having trunks and routes setup. I get a "cannot be completed as dialed. Here is the logfile for the call. Any thoughts?

[2018-06-22 00:41:07] VERBOSE[2401] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘192.168.1.106’
[2018-06-22 00:41:07] VERBOSE[2401] netsock2.c: Using SIP RTP Audio TOS bits 184
[2018-06-22 00:41:07] VERBOSE[2401] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2018-06-22 00:41:07] VERBOSE[2401] netsock2.c: Using SIP RTP Audio CoS mark 5
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [9106425168@from-internal:1] Macro(“PJSIP/5-00000003”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:1] Set(“PJSIP/5-00000003”, “TOUCH_MONITOR=1529628067.6”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:2] Set(“PJSIP/5-00000003”, “AMPUSER=5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“PJSIP/5-00000003”, “0?report”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:4] ExecIf(“PJSIP/5-00000003”, “1?Set(REALCALLERIDNUM=5)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:5] Set(“PJSIP/5-00000003”, “AMPUSER=5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:6] GotoIf(“PJSIP/5-00000003”, “0?limit”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:7] Set(“PJSIP/5-00000003”, “AMPUSERCIDNAME=5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:8] ExecIf(“PJSIP/5-00000003”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:9] GotoIf(“PJSIP/5-00000003”, “1?report”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx_builtins.c: Goto (macro-user-callerid,s,16)
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:16] NoOp(“PJSIP/5-00000003”, “Macro Depth is 1”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:17] GotoIf(“PJSIP/5-00000003”, “1?report2:macroerror”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx_builtins.c: Goto (macro-user-callerid,s,18)
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:18] GotoIf(“PJSIP/5-00000003”, “1?continue”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx_builtins.c: Goto (macro-user-callerid,s,37)
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:37] Set(“PJSIP/5-00000003”, “CALLERID(number)=5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:38] Set(“PJSIP/5-00000003”, “CALLERID(name)=5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:39] GotoIf(“PJSIP/5-00000003”, “0?cnum”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:40] Set(“PJSIP/5-00000003”, “CDR(cnam)=5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:41] Set(“PJSIP/5-00000003”, “CDR(cnum)=5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-user-callerid:42] Set(“PJSIP/5-00000003”, “CHANNEL(language)=en”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [9106425168@from-internal:2] Gosub(“PJSIP/5-00000003”, “sub-record-check,s,1(out,9106425168,dontcare)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:1] GotoIf(“PJSIP/5-00000003”, “0?initialized”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:2] Set(“PJSIP/5-00000003”, “__REC_STATUS=INITIALIZED”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:3] Set(“PJSIP/5-00000003”, “NOW=1529628067”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:4] Set(“PJSIP/5-00000003”, “__DAY=22”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:5] Set(“PJSIP/5-00000003”, “__MONTH=06”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:6] Set(“PJSIP/5-00000003”, “__YEAR=2018”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:7] Set(“PJSIP/5-00000003”, “__TIMESTR=20180622-004107”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:8] Set(“PJSIP/5-00000003”, “__FROMEXTEN=5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:9] Set(“PJSIP/5-00000003”, “__MON_FMT=wav”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:10] NoOp(“PJSIP/5-00000003”, “Recordings initialized”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:11] ExecIf(“PJSIP/5-00000003”, “0?Set(ARG3=dontcare)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:12] Set(“PJSIP/5-00000003”, “REC_POLICY_MODE_SAVE=”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:13] ExecIf(“PJSIP/5-00000003”, “0?Set(REC_STATUS=NO)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:14] GotoIf(“PJSIP/5-00000003”, “3?checkaction”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx_builtins.c: Goto (sub-record-check,s,17)
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@sub-record-check:17] GotoIf(“PJSIP/5-00000003”, “1?sub-record-check,out,1”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx_builtins.c: Goto (sub-record-check,out,1)
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [out@sub-record-check:1] NoOp(“PJSIP/5-00000003”, “Outbound Recording Check from 5 to 9106425168”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [out@sub-record-check:2] Set(“PJSIP/5-00000003”, “RECMODE=dontcare”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [out@sub-record-check:3] ExecIf(“PJSIP/5-00000003”, “1?Goto(routewins)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx_builtins.c: Goto (sub-record-check,out,7)
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [out@sub-record-check:7] Gosub(“PJSIP/5-00000003”, “recordcheck,1(dontcare,out,9106425168)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/5-00000003”, “Starting recording check against dontcare”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/5-00000003”, “dontcare”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [recordcheck@sub-record-check:3] Return(“PJSIP/5-00000003”, “”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [out@sub-record-check:8] Return(“PJSIP/5-00000003”, “”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [9106425168@from-internal:3] ExecIf(“PJSIP/5-00000003”, “0 ?Set(CDR(accountcode)=)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [9106425168@from-internal:4] Set(“PJSIP/5-00000003”, “MOHCLASS=default”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [9106425168@from-internal:5] Set(“PJSIP/5-00000003”, “_NODEST=”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [9106425168@from-internal:6] Macro(“PJSIP/5-00000003”, “dialout-trunk,1,106425168,off”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:1] Set(“PJSIP/5-00000003”, “DIAL_TRUNK=1”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:2] GosubIf(“PJSIP/5-00000003”, “0?sub-pincheck,s,1()”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:3] ExecIf(“PJSIP/5-00000003”, “0?Set(CALLERID(num)=5)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:4] GotoIf(“PJSIP/5-00000003”, “0?disabletrunk,1”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:5] Set(“PJSIP/5-00000003”, “DIAL_NUMBER=106425168”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:6] Set(“PJSIP/5-00000003”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:7] Set(“PJSIP/5-00000003”, “OUTBOUND_GROUP=OUT_1”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:8] Set(“PJSIP/5-00000003”, “DIAL_TRUNK_OPTIONS=T”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:9] GotoIf(“PJSIP/5-00000003”, “0?nomax”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:10] GotoIf(“PJSIP/5-00000003”, “0?chanfull”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:11] GotoIf(“PJSIP/5-00000003”, “0?skipoutcid”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:12] Macro(“PJSIP/5-00000003”, “outbound-callerid,1”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:1] NoOp(“PJSIP/5-00000003”, “5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:2] NoOp(“PJSIP/5-00000003”, “”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:3] NoOp(“PJSIP/5-00000003”, “off”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:4] ExecIf(“PJSIP/5-00000003”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:5] ExecIf(“PJSIP/5-00000003”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:6] ExecIf(“PJSIP/5-00000003”, “1?Set(REALCALLERIDNUM=5)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:7] GotoIf(“PJSIP/5-00000003”, “1?normcid”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx_builtins.c: Goto (macro-outbound-callerid,s,11)
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:11] Set(“PJSIP/5-00000003”, “USEROUTCID=”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:12] Set(“PJSIP/5-00000003”, “EMERGENCYCID=”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:13] Set(“PJSIP/5-00000003”, “TRUNKOUTCID=”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:14] GotoIf(“PJSIP/5-00000003”, “1?trunkcid”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx_builtins.c: Goto (macro-outbound-callerid,s,19)
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:19] ExecIf(“PJSIP/5-00000003”, “0?Set(CALLERID(all)=)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:20] ExecIf(“PJSIP/5-00000003”, “0?Set(CALLERID(all)=)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:21] ExecIf(“PJSIP/5-00000003”, “0?Set(CALLERID(all)=)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:22] ExecIf(“PJSIP/5-00000003”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:23] ExecIf(“PJSIP/5-00000003”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:24] Set(“PJSIP/5-00000003”, “CDR(outbound_cnum)=5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-outbound-callerid:25] Set(“PJSIP/5-00000003”, “CDR(outbound_cnam)=5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:13] GosubIf(“PJSIP/5-00000003”, “0?sub-flp-1,s,1()”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:14] Set(“PJSIP/5-00000003”, “OUTNUM=106425168”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:15] Set(“PJSIP/5-00000003”, “custom=SIP/callcentric”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:16] ExecIf(“PJSIP/5-00000003”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:17] ExecIf(“PJSIP/5-00000003”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:18] Macro(“PJSIP/5-00000003”, “dialout-trunk-predial-hook,”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“PJSIP/5-00000003”, “”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:19] GotoIf(“PJSIP/5-00000003”, “0?skipcrm”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:20] Set(“PJSIP/5-00000003”, “__CRM_DIRECTION=OUTBOUND”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:21] Set(“PJSIP/5-00000003”, “__CRM_DESTINATION=106425168”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:22] Set(“PJSIP/5-00000003”, “__CRM_SOURCE=5”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:23] AGI(“PJSIP/5-00000003”, “sangomacrm.agi”) in new stack
[2018-06-22 00:41:07] VERBOSE[13061][C-00000003] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] res_agi.c: <PJSIP/5-00000003>AGI Script sangomacrm.agi completed, returning 0
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:24] Set(“PJSIP/5-00000003”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:25] NoOp(“PJSIP/5-00000003”, “CRM Finished”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:26] GotoIf(“PJSIP/5-00000003”, “0?bypass,1”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:27] ExecIf(“PJSIP/5-00000003”, “1?Set(CONNECTEDLINE(num,i)=106425168)”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:28] ExecIf(“PJSIP/5-00000003”, “1?Set(CONNECTEDLINE(name,i)=CID:5)”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf(“PJSIP/5-00000003”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)5)”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:30] GotoIf(“PJSIP/5-00000003”, “0?customtrunk”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:31] Dial(“PJSIP/5-00000003”, “SIP/callcentric/106425168,300,Tb(func-apply-sipheaders^s^1)”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] netsock2.c: Using SIP RTP TOS bits 184
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] netsock2.c: Using SIP RTP CoS mark 5
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] app_stack.c: SIP/callcentric-00000003 Internal Gosub(func-apply-sipheaders,s,1) start
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/callcentric-00000003”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp(“SIP/callcentric-00000003”, “Applying SIP Headers to channel”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@func-apply-sipheaders:3] Set(“SIP/callcentric-00000003”, “SIPHEADERKEYS=”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@func-apply-sipheaders:4] While(“SIP/callcentric-00000003”, “0”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] app_while.c: Jumping to priority 8
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] pbx.c: Executing [s@func-apply-sipheaders:9] Return(“SIP/callcentric-00000003”, “”) in new stack
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] app_stack.c: Spawn extension (from-pstn, 9106425168, 1) exited non-zero on ‘SIP/callcentric-00000003’
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] app_stack.c: SIP/callcentric-00000003 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] app_dial.c: Called SIP/callcentric/106425168
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] app_dial.c: SIP/callcentric-00000003 is making progress passing it to PJSIP/5-00000003
[2018-06-22 00:41:08] VERBOSE[13061][C-00000003] app_dial.c: SIP/callcentric-00000003 is making progress passing it to PJSIP/5-00000003
[2018-06-22 00:41:09] VERBOSE[13061][C-00000003] app_dial.c: SIP/callcentric-00000003 is making progress passing it to PJSIP/5-00000003
[2018-06-22 00:41:11] VERBOSE[13061][C-00000003] app_dial.c: SIP/callcentric-00000003 is making progress passing it to PJSIP/5-00000003
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] app_macro.c: Spawn extension (macro-dialout-trunk, s, 31) exited non-zero on ‘PJSIP/5-00000003’ in macro ‘dialout-trunk’
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Spawn extension (from-internal, 9106425168, 6) exited non-zero on ‘PJSIP/5-00000003’
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/5-00000003”, “hangupcall”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/5-00000003”, “1?theend”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/5-00000003”, “0?Set(CDR(recordingfile)=)”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/5-00000003”, "SIP/callcentric-00000003 monior file= ") in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-hangupcall:5] AGI(“PJSIP/5-00000003”, “attendedtransfer-rec-restart.php,SIP/callcentric-00000003,”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] res_agi.c: <PJSIP/5-00000003>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@macro-hangupcall:6] Hangup(“PJSIP/5-00000003”, “”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/5-00000003’ in macro ‘hangupcall’
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/5-00000003’
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] app_stack.c: PJSIP/5-00000003 Internal Gosub(crm-hangup,s,1) start
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@crm-hangup:1] NoOp(“PJSIP/5-00000003”, “Sending Hangup to CRM”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@crm-hangup:2] NoOp(“PJSIP/5-00000003”, “HANGUP CAUSE: 127”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@crm-hangup:3] ExecIf(“PJSIP/5-00000003”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@crm-hangup:4] NoOp(“PJSIP/5-00000003”, “MASTER CHANNEL: 1529628067.6 = 1529628067.6”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@crm-hangup:5] GotoIf(“PJSIP/5-00000003”, “0?return”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@crm-hangup:6] Set(“PJSIP/5-00000003”, “__CRM_HANGUP=1”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@crm-hangup:7] AGI(“PJSIP/5-00000003”, “sangomacrm.agi”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] res_agi.c: <PJSIP/5-00000003>AGI Script sangomacrm.agi completed, returning 0
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] pbx.c: Executing [s@crm-hangup:8] Return(“PJSIP/5-00000003”, “”) in new stack
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/5-00000003’
[2018-06-22 00:41:12] VERBOSE[13061][C-00000003] app_stack.c: PJSIP/5-00000003 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

Posts: 5

Participants: 2

Read full topic

↧

CRM Module - what records does it exactly ingest?

June 21, 2018, 6:29 pm
$
0
0

@kgermann wrote:

I know that if a record (lead, account, contact) exists it will attach call records to it, but I want to confirm that even if they don’t exist, the records are still pushed to my CRM and stored there.

I’m trying to build some custom call reporting, so this would be a required feature.

Is there a way I could modify the module to push CDRs to the CRM even if a lead/contact didn’t exist?

Posts: 1

Participants: 1

Read full topic

↧
↧

Cisco 8961 SIP help

June 22, 2018, 10:08 am
$
0
0

@threeeye wrote:

Hi guys,
I have a problem with Cisco 8961
I got it to register just fine, my problem is with the softkeys and features…

  1. I need the DND key on the bottom row
  2. I need a BLF (or 2) in the right lines (with Speed Dial, Pickup and subscription (red and green led))
  3. The button for the VM gives me a message “Messages not available”

Here is my SEP:

<?xml version="1.0" ?>
<device>
  <deviceProtocol>SIP</deviceProtocol>
  <sshUserId>admin</sshUserId>
  <sshPassword>cisco</sshPassword>
  <devicePool>
    <dateTimeSetting>
			<name>US Eastern Standard Time</name>
			<dateTemplate>M/D/YYA</dateTemplate>
			<timeZone>US Eastern Standard Time</timeZone>
			<olsonTimeZone>US Eastern Standard Time</olsonTimeZone>
      <ntps>
        <ntp>
          <name>GW-IP-ADDRESS</name>
          <ntpMode>Unicast</ntpMode>
        </ntp>
      </ntps>
    </dateTimeSetting>
    <callManagerGroup>
      <members>
        <member priority="0">
          <callManager>
            <processNodeName>FREEPBX_SERVER</processNodeName>
            <ports>
              <sipPort>5160</sipPort>
            </ports>
          </callManager>
        </member>
      </members>
    </callManagerGroup>
  </devicePool>
  <sipProfile>
    <natEnabled>false</natEnabled>
    <natAddress></natAddress>
    <sipProxies>
        <backupProxy>USECALLMANAGER</backupProxy>
        <backupProxyPort>5160</backupProxyPort>
        <emergencyProxy>USECALLMANAGER</emergencyProxy>
        <emergencyProxyPort>5160</emergencyProxyPort>
        <outboundProxy></outboundProxy>
        <outboundProxyPort></outboundProxyPort>
        <registerWithProxy>true</registerWithProxy>
     </sipProxies>
    <preferredCodec></preferredCodec>
    <phoneLabel>NAME</phoneLabel>
    <sipLines>
      <line button="1">
        <featureID>9</featureID>
        <featureLabel>EXT</featureLabel>
        <proxy>USECALLMANAGER</proxy>
        <port>5160</port>
        <name>EXT</name>
        <displayName>NAME</displayName>
        <autoAnswer>
          <autoAnswerEnabled>2</autoAnswerEnabled>
        </autoAnswer>
        <callWaiting>3</callWaiting>
        <authName>EXT</authName>
        <authPassword>EXT_PASSWORD</authPassword>
        <sharedLine>false</sharedLine>
        <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
        <messagesNumber>*97</messagesNumber>
        <ringSettingIdle>4</ringSettingIdle>
        <ringSettingActive>5</ringSettingActive>
        <contact>EXT</contact>
        <forwardCallInfoDisplay>
          <callerName>true</callerName>
          <callerNumber>true</callerNumber>
          <redirectedNumber>true</redirectedNumber>
          <dialedNumber>true</dialedNumber>
        </forwardCallInfoDisplay>
      </line>
      <line button="2">
        <featureID></featureID>
        <featureLabel></featureLabel>
        <featureOptionMask></featureOptionMask>
        <speedDialNumber></speedDialNumber>
      </line>
      <line button="3">
        <featureID>21</featureID>
        <featureLabel>SOME_NAME</featureLabel>
        <featureOptionMask>1</featureOptionMask>
        <speedDialNumber>EXT</speedDialNumber>
      </line>
    </sipLines>
    <dialTemplate>dialplan.xml</dialTemplate>
    <softKeyFile>softkey.xml</softKeyFile>
  </sipProfile>
  <featurePolicyFile>feature.xml</featurePolicyFile>
  <userLocale>
    <name></name>
    <langCode></langCode>
  </userLocale>
  <networkLocale></networkLocale>
  <networkLocaleInfo>
    <name></name>
  </networkLocaleInfo>
  <vendorConfig>
	<wirelessMicRegion>0</wirelessMicRegion>
	<webAccess>0</webAccess>
	<moreKeyReversionTimer>5</moreKeyReversionTimer>
	<g722CodecSupport>0</g722CodecSupport>
	<lldpAssetId></lldpAssetId>
	<powerPriority>0</powerPriority>
	<displayRefreshRate>0</displayRefreshRate>
	<useEnblocDialing>1</useEnblocDialing>
  </vendorConfig>
  <transportLayerProtocol>2</transportLayerProtocol>
</device>

Here is my feature.xml:

<featurePolicy name="Feature Policy">
  <versionStamp>a64c9b2e-b1fe-4781-ba89-7f8574012eb8</versionStamp>
  <featureDef name="Forward All">
    <id>1</id>
    <enable>true</enable>
  </featureDef>
  <featureDef name="Park">
    <id>2</id>
    <enable>true</enable>
  </featureDef>
  <featureDef name="Speed Dial">
    <id>5</id>
    <enable>true</enable>
  </featureDef>
  <featureDef name="Call Back">
    <id>6</id>
    <enable>true</enable>
  </featureDef>
  <featureDef name="Redial">
    <id>7</id>
    <enable>true</enable>
  </featureDef>
  <featureDef name="Barge">
    <id>8</id>
    <enable>false</enable>
  </featureDef>
</featurePolicy>

Here is my softket.xml:

<softKeyCfg>
<versionStamp>11cdf71b-e9bc-4559-be88-94a266766601</versionStamp>
<typeSoftKey>
<softKeyDef keyID="Redial">
<tag>1</tag>
<eventID>1</eventID>
<helpID>301</helpID>
</softKeyDef>
<softKeyDef keyID="NewCall">
<tag>2</tag>
<eventID>2</eventID>
<helpID>302</helpID>
</softKeyDef>
<softKeyDef keyID="Hold">
<tag>3</tag>
<eventID>3</eventID>
<helpID>303</helpID>
</softKeyDef>
<softKeyDef keyID="Trnsfer">
<tag>4</tag>
<eventID>4</eventID>
<helpID>304</helpID>
</softKeyDef>
<softKeyDef keyID="CfwdAll">
<tag>5</tag>
<eventID>5</eventID>
<helpID>305</helpID>
</softKeyDef>
<softKeyDef keyID="CfwdBusy">
<tag>6</tag>
<eventID>6</eventID>
<helpID>306</helpID>
</softKeyDef>
<softKeyDef keyID="CfwdNoAnswer">
<tag>7</tag>
<eventID>7</eventID>
<helpID>307</helpID>
</softKeyDef>
// back
<softKeyDef keyID="&lt;&lt;">
<tag>8</tag>
<eventID>8</eventID>
<helpID>308</helpID>
</softKeyDef>
<softKeyDef keyID="EndCall">
<tag>9</tag>
<eventID>9</eventID>
<helpID>309</helpID>
</softKeyDef>
<softKeyDef keyID="Resume">
<tag>10</tag>
<eventID>10</eventID>
<helpID>310</helpID>
</softKeyDef>
<softKeyDef keyID="Answer">
<tag>11</tag>
<eventID>11</eventID>
<helpID>311</helpID>
</softKeyDef>
<softKeyDef keyID="Info">
<tag>12</tag>
<eventID>12</eventID>
<helpID>312</helpID>
</softKeyDef>
<softKeyDef keyID="Confrn">
<tag>13</tag>
<eventID>13</eventID>
<helpID>313</helpID>
</softKeyDef>
<softKeyDef keyID="Park">
<tag>14</tag>
<eventID>14</eventID>
<helpID>314</helpID>
</softKeyDef>
<softKeyDef keyID="Join">
<tag>15</tag>
<eventID>15</eventID>
<helpID>315</helpID>
</softKeyDef>
<softKeyDef keyID="MeetMe">
<tag>16</tag>
<eventID>16</eventID>
<helpID>316</helpID>
</softKeyDef>
<softKeyDef keyID="PickUp">
<tag>17</tag>
<eventID>17</eventID>
<helpID>317</helpID>
</softKeyDef>
<softKeyDef keyID="GPickUp">
<tag>18</tag>
<eventID>18</eventID>
<helpID>318</helpID>
</softKeyDef>
// remove last conference party
<softKeyDef keyID="RmLstC">
<tag>57</tag>
<eventID>19</eventID>
<helpID>319</helpID>
</softKeyDef>
<softKeyDef keyID="Barge">
<tag>67</tag>
<eventID>21</eventID>
<helpID>321</helpID>
</softKeyDef>
<softKeyDef keyID="DirTrfr">
<tag>77</tag>
<eventID>28</eventID>
<helpID>328</helpID>
</softKeyDef>
<softKeyDef keyID="Select">
<tag>78</tag>
<eventID>29</eventID>
<helpID>329</helpID>
</softKeyDef>
<softKeyDef keyID="ConfList">
<tag>79</tag>
<eventID>30</eventID>
<helpID>330</helpID>
</softKeyDef>
<softKeyDef keyID="VidMode">
<tag>88</tag>
<eventID>33</eventID>
<helpID>333</helpID>
</softKeyDef>
<softKeyDef keyID="ImmDiv">
<tag>59</tag>
<eventID>65</eventID>
<helpID>365</helpID>
</softKeyDef>
<softKeyDef keyID="Intrcpt">
<tag>60</tag>
<eventID>66</eventID>
<helpID>366</helpID>
</softKeyDef>
<softKeyDef keyID="SetWtch">
<tag>61</tag>
<eventID>67</eventID>
<helpID>367</helpID>
</softKeyDef>
<softKeyDef keyID="TrnsfVM">
<tag>62</tag>
<eventID>68</eventID>
<helpID>368</helpID>
</softKeyDef>
<softKeyDef keyID="DND">
<tag>63</tag>
<eventID>69</eventID>
<helpID>369</helpID>
</softKeyDef>
<softKeyDef keyID="DivAll">
<tag>64</tag>
<eventID>70</eventID>
<helpID>370</helpID>
</softKeyDef>
</typeSoftKey>
<softKeySets>
<softKeySet id="On Hook">
<softKey keyID="Redial"></softKey>
<softKey keyID="NewCall"></softKey>
<softKey keyID="Intrcpt"></softKey>
<softKey keyID="DND"></softKey>
<softKey keyID="CfwdAll"></softKey>
<softKey keyID="DivAll"></softKey>
<softKey keyID="SetWtch"></softKey>
</softKeySet>
<softKeySet id="Connected">
<softKey keyID="Hold"></softKey>
<softKey keyID="EndCall"></softKey>
<softKey keyID="Intrcpt"></softKey>
<softKey keyID="ImmDiv"></softKey>
<softKey keyID="Confrn"></softKey>
<softKey keyID="ConfList"></softKey>
<softKey keyID="Trnsfer"></softKey>
<softKey keyID="TrnsfVM"></softKey>
<softKey keyID="DND"></softKey>
<softKey keyID="Select"></softKey>
<softKey keyID="Join"></softKey>
<softKey keyID="DirTrfr"></softKey>
<softKey keyID="RmLstC"></softKey>
<softKey keyID="Park"></softKey>
<softKey keyID="DivAll"></softKey>
<softKey keyID="SetWtch"></softKey>
</softKeySet>
<softKeySet id="On Hold">
<softKey keyID="Resume"></softKey>
<softKey keyID="NewCall"></softKey>
<softKey keyID="Intrcpt"></softKey>
<softKey keyID="TrnsfVM"></softKey>
<softKey keyID="ImmDiv"></softKey>
<softKey keyID="DND"></softKey>
<softKey keyID="Select"></softKey>
<softKey keyID="Join"></softKey>
<softKey keyID="DirTrfr"></softKey>
<softKey keyID="SetWtch"></softKey>
</softKeySet>
<softKeySet id="Ring In">
<softKey keyID="Answer"></softKey>
<softKey keyID="ImmDiv"></softKey>
<softKey keyID="CfwdBusy"></softKey>
<softKey keyID="TrnsfVM"></softKey>
<softKey keyID="DivAll"></softKey>
<softKey keyID="DND"></softKey>
<softKey keyID="SetWtch"></softKey>
</softKeySet>
<softKeySet id="Off Hook">
<softKey keyID="Redial"></softKey>
<softKey keyID="EndCall"></softKey>
<softKey keyID="Intrcpt"></softKey>
<softKey keyID="PickUp"></softKey>
<softKey keyID="GPickUp"></softKey>
<softKey keyID="CfwdAll"></softKey>
<softKey keyID="MeetMe"></softKey>
</softKeySet>
<softKeySet id="Connected Transfer">
<softKey keyID="Intrcpt"></softKey>
<softKey keyID="EndCall"></softKey>
<softKey keyID="Trnsfer"></softKey>
</softKeySet>
<softKeySet id="Digits After First">
<softKey keyID="&lt;&lt;"></softKey>
<softKey keyID="EndCall"></softKey>
<softKey keyID="Intrcpt"></softKey>
</softKeySet>
<softKeySet id="Connected Conference">
<softKey keyID="Intrcpt"></softKey>
<softKey keyID="EndCall"></softKey>
<softKey keyID="Confrn"></softKey>
</softKeySet>
<softKeySet id="Ring Out">
<softKey keyID="EndCall"></softKey>
<softKey keyID="Intrcpt"></softKey>
</softKeySet>
<softKeySet id="Off Hook With Feature">
<softKey keyID="Redial"></softKey>
<softKey keyID="EndCall"></softKey>
<softKey keyID="Intrcpt"></softKey>
</softKeySet>
<softKeySet id="Remote In Use">
<softKey keyID="Barge"></softKey>
<softKey keyID="NewCall"></softKey>
<softKey keyID="cBarge"></softKey>
</softKeySet>
</softKeySets>
</softKeyCfg>

Thanks for the help

Posts: 1

Participants: 1

Read full topic

↧

DND in the middle of a call

June 22, 2018, 12:01 pm
$
0
0

@brianbkis wrote:

Using a Grandstream GXP2160, while in the middle of a call if I turn on DND (using a softkey programmed for *76) it puts the call on hold and enables DND. Dialing *76 or *78 manually doesn’t work at all…wondering if that is a DTMF issue, but haven’t had any luck with different canreinvite settings. Is it possible to enable DND mid-call without putting the current user on hold? Thanks!

Posts: 4

Participants: 3

Read full topic

↧
Search

Sipp dtmf tones not recognised

June 22, 2018, 12:48 pm
$
0
0

@Bobd22 wrote:

Hello,

I have been having a rough time trying to send dtmf with sipp. At first I was using sippy_cup but decided I’d try this in sipp. I was having similar issues with sippy_cup. I have been trying to replay pcap files and the included dtmf tones in the /pcap directory for sipp and some captures I’ve got with tcpdump. I built sipp from source with pcap support SIPp v3.5.1-PCAP-RTPSTREAM. I’m running FreePBX 14.0.3.6 from raspbx on a rasberry pi for testing. I’m running sipp on the same host as FreePBX also.

Goal: Test ivr with 5-6 dtmf tones for load and errors.

In the cdr reports I always see sipp calling from the destination “s [from-trunk]” in my cdr reports. I can see the dtmf tones in the full log and asterisk cli like below. I have also tried all the different dtfm modes in the Settings>Advanced Settings and the trunk details.

[2018-06-22 13:06:32] DTMF[9942][C-00000028]: channel.c:4040 __ast_read: DTMF end ‘1’ received on SIP/127.0.1.1-00000028, duration 0 ms
[2018-06-22 13:06:32] DTMF[9942][C-00000028]: channel.c:4099 __ast_read: DTMF end accepted without begin ‘1’ on
SIP/127.0.1.1-00000028
[2018-06-22 13:06:32] DTMF[9942][C-00000028]: channel.c:4110 __ast_read: DTMF end passthrough ‘1’ on SIP/127.0.1.1-00000028
[2018-06-22 13:06:35] DTMF[9959][C-00000029]: channel.c:4040 __ast_read: DTMF end ‘1’ received on SIP/127.0.1.1-00000029, duration 0 ms
[2018-06-22 13:06:35] DTMF[9959][C-00000029]: channel.c:4099 __ast_read: DTMF end accepted without begin ‘1’ on
SIP/127.0.1.1-00000029

Scenario below

<?xml version="1.0" encoding="ISO-8859-1" ?> <![CDATA[ 
  INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
  To: [service] <sip:[service]@[remote_ip]:[remote_port]>
  Call-ID: [call_id]
  CSeq: 1 INVITE
  Contact: sip:sipp@[local_ip]:[local_port]
  Max-Forwards: 70
  Subject: Performance Test
  Content-Type: application/sdp
  Content-Length: [len]

  v=0
  o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
  s=-
  c=IN IP[local_ip_type] [local_ip]
  t=0 0
  m=audio [auto_media_port] RTP/AVP 8 101
  a=rtpmap:8 PCMA/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-11,16

]]>

                                                                                                     [11/72]
<![CDATA[ 
  ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
  To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  Call-ID: [call_id]
  CSeq: 1 ACK
  Contact: sip:sipp@[local_ip]:[local_port]
  Max-Forwards: 70
  Subject: Performance Test
  Content-Length: 0

]]>
<![CDATA[ 
  BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
  To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  Call-ID: [call_id]
  CSeq: 2 BYE
  Contact: sip:sipp@[local_ip]:[local_port]
  Max-Forwards: 70
  Subject: Performance Test
  Content-Length: 0

]]>

I’ve tried this in a scenario also. Thinking I needed a pause between the tones.

I have been playing some pcaps that I got via tcpdump and the included dtmf tones in the pcap directory. I can see the dtfm tones in the call flow in wireshark and hear them. I also separated both legs of the call to try just the part from the trunk with the tones. I got the same results as having both legs of the call in the pcap. I even tried some from wiresharks site and got some results. I can see asterisk responding with SayAlpha in the cdr reports and the logs.

To acheive this do I need to patch sipp with the inband dtfm patch or some other patch? I’ve tried and I can’t compile it after the patch.

How can I get sipp calling my ivr and getting the dtmf tones accepted? Am I using the wrong tool for this? Is there anything better? I was thinking about making a script to just make the calls like normal? Thanks for taking the time to read this.

Have a good day.

Posts: 1

Participants: 1

Read full topic

↧

Endpoint Manager won't see extension

June 23, 2018, 12:59 pm
$
0
0

@sentinelace wrote:

I have a new build on freepbx 14. 4 extensions. All went fine with endpoint manager except one phone. Extension 103 will not show up in endpoint manager. I have tried deleting and re-creating it with no go. When I go to add, I just have “custom” as an option and that’s it.

Posts: 1

Participants: 1

Read full topic

↧
↧

Calendar syntax details

June 23, 2018, 10:54 pm
$
0
0

@Mypbx wrote:

I’m trying to implement 2 calendars in my call routing:

  • Agenda with National pubic holidays
  • Agenda with Override regular opening hours

Via the statement IfThen I would like to control this, but I would like to have the calendar syntax to be used in the IfThen statement. This is not clear to me and I can’t find it. What is being tested (event in calendar present ?) and which results can I expect ?

And ofcourse details about the syntax

Thanks in advance

Posts: 1

Participants: 1

Read full topic

↧

High CPU usage in idle: recordingreport

June 24, 2018, 1:19 pm
$
0
0

@vegbrasil wrote:

Hi there,

After upgrading a few modules since a few days ago, I’m seeing insane high loads from a cron job that runs every 15 minutes:

/usr/sbin/fwconsole recordingreport -s -q

It finishes in 10 minutes or so and returns (obviously) in the next 5 minutes with the high CPU usage - repeating for ~10 days now.

I’ve tried to run this command with verbose argument (-vvv) and there’s no output to help diagnose the problem.

I have a few questions:

  • What this command does? I do have the commercial module Call Recording Report, is it related?
  • Is it safe to comment-out this cronjob for now?

My setup is running the FreePBX distro with FreePBX 14.0.3.6, all modules updated.

Thanks!

Posts: 1

Participants: 1

Read full topic

↧

Blind Transfer to VM (works but not 100%)

June 24, 2018, 10:55 pm
$
0
0

@rsarceno wrote:

FreePBX 13.0.195.4
PBX Firmware:10.13.66-22
Asterisk Version: 13.19.1

  • I’m getting the following result using Sangoma s700 (ext 101), Yealink T27P (ext 102) and Sangoma s500 (ext 99998)
  • Blind transfer to VM works on some extensin but not to all all ext with VM (eg it works if I transfer to ##*99997 but not if I transfer to ##*101)
  • Blind Transfer (not voicemail) to any extension works (##9998 or ##101)

Since I can transfer to 99998 but not 101, I’ve concentrated my troubleshooting comparing these two ext and they are almost identical. I

I updated the firmware on S700 to 2.0.4.53 but it didn’t help.

Appreciate any information

Posts: 1

Participants: 1

Read full topic

↧
Viewing all 12616 articles
Browse latest View live
Search
<script src="https://jsc.adskeeper.com/r/s/rssing.com.1596347.js" async> </script>