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Prepend non-numeric character to Outbound route

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@arielgrin wrote:

Dear all: As I'm using an SPA400 and I need to choose which line to use depending on the outbound route matched, I need to prepend a non-numeric character (L1, L2, L3 or L4, depending the line to be used) to the outbound route, but the prepend field only accepts numerals. If I enter L1 it gets changed to 1, L2 to 2 and so on. Is there a way to prepend a non-numeric character to the dialed number?

I see that maybe I could use "Outbound Dial Prefix" option of the trunk, but to use each line I would need to create 4 different trunks, all pointing to the same SPA400, and I don't think that it very intuitive.

I thought about using custom contexts for the different outbound routes, but I'm not sure how to move forward. Should I delete my outbound routes that are defined through the Outbound Routes menu and manually create the custom context? Or leave the outbound routes in place and create the custom contexts anyway?

Thanks, Ariel.

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Backup config and restore on new server

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@robertkwild wrote:

hi all,

i have installed freepbx with extensions, trunk to my sip provider, xmpp and i just would like to know how to back it up and restore it on a brand new server, so all my extensions and trunk to sip provider,xmpp work out the box when i restore it on the new server

many thanks,

rob

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Sporlac DS

Error While upgrading to FreePBX 14 Distro

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@danemgroup wrote:

I upgraded to Freepbx 14. After that i cannot login to GUI. CLI works fine.
Found one module Time conditions disabled with fwconsole ma list.
When i tried upgrading it gives an error as mentioned from its dependecy cel.

[root@danemqtrmain ~]# fwconsole ma enable cel
The following error(s) occured:
- Module cel cannot be enabled
[root@danemqtrmain ~]# fwconsole ma enable timeconditions
The following error(s) occured:
- Module timeconditions cannot be enabled
[root@danemqtrmain ~]# fwconsole ma install cel
Creating cel if needed..

[Exception]
Can not create cel table::

ma [-f|--force] [-d|--debug] [--edge] [--color] [--skipchown] [-e|--autoenable] [--skipdisabled] [--snapshot SNAPSHOT] [--format FORMAT] [-R|--repo REPO] [-t|--tag TAG] [--onlystdout] [--willupdate] [--securityonly] [--sendemails] [--] []...

Please help me out guys

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Connecting two FreePBX boxes with a SIP Trunk

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@kwriley87 wrote:

Hello

I have Server A and Server B. I want to configure a SIP trunk between the two servers so that all outbound calls on Server A route through Server B which is connected to my carrier.

I've got a SIP trunk registered between them both, but when I attempt a call, it never hits Server B. Server A provides the following output:

-- Executing [s@macro-dialout-trunk-predial-hook:1] SIPAddHeader("SIP/652-00001aec", "P-Asserted-Identity:sip:4698503805@bt.voipdnsservers.com") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:2] MacroExit("SIP/652-00001aec", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/652-00001aec", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/652-00001aec", "1?Set(CONNECTEDLINE(num,i)=2146749682)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/652-00001aec", "1?Set(CONNECTEDLINE(name,i)=CID:4698503805)") in new stack
-- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/652-00001aec", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:22] Dial("SIP/652-00001aec", "SIP/Call-Router-A/2146749682,300,Tt") in new stack
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/652-00001aec", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/652-00001aec", "0?continue,1:s-CONGESTION,1") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/652-00001aec", "RC=21") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/652-00001aec", "21,1") in new stack
-- Executing [21@macro-dialout-trunk:1] Goto("SIP/652-00001aec", "continue,1") in new stack
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/652-00001aec", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] Set("SIP/652-00001aec", "CALLERID(number)=652") in new stack
-- Executing [2146749682@restrictedroute-9bf31c7ff062936a96d3c8bd1f8f2ff3:7] Macro("SIP/652-00001aec", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/652-00001aec", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/652-00001aec", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/652-00001aec", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/652-00001aec", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- Playing 'all-circuits-busy-now.ulaw' (language 'en')
-- Playing 'pls-try-call-later.ulaw' (language 'en')

Here are my trunk config's for both servers..

Server A
type=friend
context=from-internal
qualify=yes
host=IP ADDRESS OF SERVER B
insecure=invite
allow=all

Server B
type=friend
context=from-internal
qualify=yes
insecure=port,invite
host=CARRIER IP
fromdomain=CARRIER IP
disallow=all
allow=g729,ulaw

Can someone please help me understand what is wrong with my trunk config?

Thank you so much!

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Return Caller ID

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@tigger1197 wrote:

I just reinstalled the latest distro 14.0 and this time when a person is calling an extension this is happening

ext 100 - room A
ext 200 - room B

ext 100 calls 200. 200 receives caller id of "room A" but when the call is answered ext 100 doesn't see the display name of 200. The display name only shows up on one direction.

It used to work both ways in my last install. How do I fix it so the caller id/display name is showed in both directions?

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Send hook flash to a PSTN line connected through a SIP gateway

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@arielgrin wrote:

Hi all. I know that a hook flash can be send through a dahdi channel but I need to send it through a SIP gateway. I know and verified that the SIP gateway correctly understands DTMF event 16, which corresponds to the hook flash, and is able to send the hook flash to the PSTN line. What I don't know and this is where I need some help, is how to generate the hookflash from a SIP phone so it gets aent to the gateway. I have read some articles but they all discuss hook flash on dahdi channels. Thanks, Ariel.

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Error when creating self signed certificate

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@westcana wrote:

I'm having a problem with self signed certificates on a Sangoma phone system 40 device.
It is running Freepbx V13.0.192.16
I have all the modules up to date.
I'm doing the setup for a new deployment and everthing is working fine so far.
I'm at the point where I am setting up secure connections for the management and UCP websites.
The first time I went to certificate managment and set the default certificate (the one that is created when Freepbx is activated).
Then went to system admin - https setup - settings and selected the certificate and clicked install.
Then I went to system admin - port management and set the admin to port 1443 and ucp to 443 and hit aupdate now. (just a note, I have tried other combinations with the same results).
When I try to connect to the site using https://mypbx.com:1443 with IE or Chrome, I get an error and can't continue to the site.
I use both of these browsers to connect to other system that I manage that use self signed certificates, so I am familiar with how to connect when the self signed certificate is used.
I have tried to generate new certificates and set them as default, etc, with the same results.
The actual error in Chrome is
"xxxxx normally uses encryption to protect your information. When Google Chrome tried to connect to xxxx this time, the website sent back unusual and incorrect credentials. This may happen when an attacker is trying to pretend to be xxxxx, or a Wi-Fi sign-in screen has interrupted the connection. Your information is still secure because Google Chrome stopped the connection before any data was exchanged.

You cannot visit xxxxx right now because the website sent scrambled credentials that Google Chrome cannot process. Network errors and attacks are usually temporary, so this page will probably work later."
I even went so far as to reboot the server, without and change.

Any ideas are welcome.
Thanks!

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*93 Call Forward Feature Code

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@cramermp wrote:

Hello all. Using FreePBX 13 (distro 10), asterisk 13.

I took a look at the following resource on the wiki, but the information at the top doesn't match the information further down, where it refers to feature code *720 which is no longer a default feature code (apparently due to conflicting with *72). It seems like it's missing some important information, and there are no examples to follow to make sure I'm properly executing the steps to use this feature code.
https://wiki.freepbx.org/display/FPG/Call+Forwarding+User+Guide#CallForwardingUserGuide-ALL(unconditional)

When I dial the *93 feature code for call forwarding, labeled (and enabled) in the system as:
Call Forward All Prompting Activate
I experience dial tone for the feature code entry, then a pause, then another dial tone. No matter what I enter at this point, it just originates a call to what I dial, whether it's an extension, an external number, whatever. I was expecting Allison to ask me for an extension, and then ask me for a number to forward to, but there's no "prompting." Even if Allison didn't prompt me for anything, I can only input a single number and it just originates a call, doesn't do anything with call forwarding.

Is there something I'm missing?

Just fyi, in case some other solution exists, we're looking to set up call forwarding for an extension while we're not at the office, using just the phone system and dial codes (the UCP would work, but we'd have to fully set up our users, get certificates, port forwarding, distribute passwords & train on interface usage, etc. and there's more red tape to that than just setting up a DID to DISA).

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How Does One Migrate the CDR Database?

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@POTShead wrote:

Hello! I am running an older FreePBX (v12.0.19, Asterisk v13.0.2) in a virtual machine and have setup a new VM with the current FreePBX (v14.0.1.1, Asterisk v13.17.0).

I'd greatly prefer to migrate the old CDR database onto the new setup, right before I go live with it (and so do by way of copying the database files, instead of exporting/importing, etc).

I'm wondering if there is something like:
copy C:\FreePBX.12\data\CDR.dbf C:\FreePBX.14\data*.*
that'll do this?

(I'll translate to SCP as necessary ; )

Cheers & thanks for any leads!

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FreePBX External Extension Oddball Port

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@mm999 wrote:

Hi everyone, my FreePBX system has been working great for quite a while, but I've always have been curious why it uses a oddball port 1024 for one of my remote extensions.

The remote office has a Linksys 3102 with both the Line (111) and PSTN trunk connected along with a Linksys phone extension (112). All extensions work fine at this location.

Maybe I missed a configuration setting somewhere but I did a quick review and all seemed to be ok. Any ideas?

Thanks in advance!

Asterisk Info:
Name/username Host D For Coma ACL Port Sta Description
103/103 192.168.0.125 D No No A 5060 OK (4 ms)
105/105 192.168.0.173 D No No A 5060 OK (9 ms)
108/108 206.116.XXX.XXX D Yes Yes A 5060 OK (47 ms)
110/110 192.168.0.173 D No No A 5060 OK (9 ms)
111/111 162.157.XXX.XXX D Yes Yes A 5060 OK (36 ms)
112/112 162.157.XXX.XXX D Yes Yes A 1024 OK (48 ms)
...

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International Dial Patterns Not Working

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@ashcortech wrote:

Hey all,

Trying to set up dial patterns for international dialing from the US.

the pattern put into the outbound route from the wizard is: 011. which seems correct.

However this will not work. I had to modify it to 01144XXXXXXXXXXX (trying to dial the UK) in order for a call to go out.

does the "." wildcard not work?

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Log other users in/out of queues

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@bigmillz wrote:

Sometimes I have to log another user in or out, and don't have time to call them and tell them to do it themselves.

*45*xxx(their extension) doesn't do it, it keeps logging me in and out instead.

Is there an easy way? If it involves running a bash script, it's not worth it to me. Less time consumed asking them to do it.

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Stopping mysqld in FreePBX 14

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@dobrosavljevic wrote:

I would like to know what the proper way to shut down the MySQL service is going forward in FreePBX 14.

In FreePBX 13 I was able to do a simple service mysqld stop/start, however in 14 that is no longer an option. Is that now handled by fwconsole stop/start or am I just missing something obvious?

Direction with this is greatly appreciated.

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LDAP user groups

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@ashleydrees wrote:

When i sync FreePBX 14 with an OpenLDAP server (on Ubuntu Linux 16.04.1) i get all the users and i get all the groups but ONLY the primary group shows up in the "User Manager" and i am unable to edit the field - is this by design?

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FreePBX Restart Issue with Cisco 7940 (SIP)

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@brandon1767 wrote:

Hey everyone,

I am having an issue regarding my Cisco 7940 and FreePBX. After my FreePBX box restarts, for some odd reason, I have to go to my Cisco 7940 and Enable then Disable NAT in the SIP Configuration menu. If I do not do this, I am unable to receive incoming calls after my FreePBX box restarts.

After the box restarts, I can't even tell if it is working without calling myself and seeing if the phone rings as it shows the connected icon on the phone.

This gets pretty annoying, especially when the box restarts without my knowledge and I am missing customer calls.

Any help would be much appreciated! Feel free to ask any questions.

Thanks,

Brandon

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One way audio when paging outdoor intercom (CyberData)

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@scstone18 wrote:

I'm using Freepbx 14.0.1.4 and I have a CyberData VoIP intercom Model 010935B. I have the outside intercom (CyberData) set to auto answer. When I page from the outside intercom station everything works great! But when I page the intercom from an extension the Cyberdata auto answers as programmed but only one way audio (No audio is being transmitted to the Cyberdata outdoor intercom) I've read about one way audio and asterisk and tried to troubleshoot. But if I page other extensions S500 phones, everything functions normally. I've changed the rtp port on the Cyberdata and still have the same issue. Any help would be very much appreciated!

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No Audio over public IP

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@sps wrote:

Hi, Everyone!

I had a question I was hoping you all could answer.
I've had a PBX online for about a year now and I have to make and receive calls over the public IP. Now, on some internet connections this isn't an issue, on others I can't hear the other end of the call. Also, Since I have a team that uses the system the same way, some of them don't have any issue at all and some of them do.

Is it something to do with NAT? Everything I've found online to remedy the situation has been confusing and totally worthless.

If you all have any idea, it would be greatly appreciated.

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Remote extension's voicemail answering calls from a Queue

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@frazzels wrote:

Hello,

I'm hoping someone can help me. I have two FreePBX 13 servers connected together using an IAX2 trunk.

Server A has a call queue which includes local Server A extensions. I would now like to add some remote extensions located on Server B to the Call queue on Server A. This works as expected however the problem happens when any of the remote extensions are busy. Rather than Server A registering that the remote extensions are busy (as it does with the local extensions), the call from the queue gets put through to the busy remote extension's voicemail. Skip busy agents is enabled on the Queue however this only works for the local extensions.

Similar posts here, here, and here. All of which say use Confirm calls however when Confirm Calls is enabled voicemails are being left on the remote extensions mailbox saying "press 1 to accept or press 2 to decline"

Disabling voicemail on the remote extensions is not possible. Also I'd rather not attached the remote extensions directly to Server A if at all possible

Thanks
Fraser

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Can't make outgoing calls

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@colinhowlin wrote:

I've set up a new installation of FreePBX and set up a trunk and 5 extensions.
Incoming calls work perfectly but calls out to any number fail with message saying "The number you have dialed is unavailable".

See Askterisk log below.

Any help would be greatly appreciated.
If you need any more info please let me know.

Colin

[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [0872910755@from-internal:1] Macro("PJSIP/100-0000007f", "user-callerid,LIMIT,EXTERNAL,") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:1] Set("PJSIP/100-0000007f", "TOUCH_MONITOR=1504520424.129") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:2] Set("PJSIP/100-0000007f", "AMPUSER=100") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("PJSIP/100-0000007f", "0?report") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("PJSIP/100-0000007f", "1?Set(REALCALLERIDNUM=100)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:5] Set("PJSIP/100-0000007f", "AMPUSER=100") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("PJSIP/100-0000007f", "0?limit") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:7] Set("PJSIP/100-0000007f", "AMPUSERCIDNAME=Colin") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:8] GotoIf("PJSIP/100-0000007f", "0?report") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:9] Set("PJSIP/100-0000007f", "AMPUSERCID=100") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:10] Set("PJSIP/100-0000007f", "__DIAL_OPTIONS=Ttr") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:11] Set("PJSIP/100-0000007f", "CALLERID(all)="Colin" <100>") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:12] GotoIf("PJSIP/100-0000007f", "0?limit") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:13] ExecIf("PJSIP/100-0000007f", "1?Set(GROUP(concurrency_limit)=100)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:14] ExecIf("PJSIP/100-0000007f", "0?Set(CHANNEL(language)=)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:15] GotoIf("PJSIP/100-0000007f", "1?continue") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (macro-user-callerid,s,29)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:29] Set("PJSIP/100-0000007f", "CALLERID(number)=100") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:30] Set("PJSIP/100-0000007f", "CALLERID(name)=Colin") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:31] GotoIf("PJSIP/100-0000007f", "0?cnum") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:32] Set("PJSIP/100-0000007f", "CDR(cnam)=Colin") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:33] Set("PJSIP/100-0000007f", "CDR(cnum)=100") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-user-callerid:34] Set("PJSIP/100-0000007f", "CHANNEL(language)=en") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [0872910755@from-internal:2] Gosub("PJSIP/100-0000007f", "sub-record-check,s,1(out,0872910755,dontcare)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:1] GotoIf("PJSIP/100-0000007f", "0?initialized") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:2] Set("PJSIP/100-0000007f", "__REC_STATUS=INITIALIZED") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:3] Set("PJSIP/100-0000007f", "NOW=1504520424") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:4] Set("PJSIP/100-0000007f", "__DAY=04") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:5] Set("PJSIP/100-0000007f", "__MONTH=09") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:6] Set("PJSIP/100-0000007f", "__YEAR=2017") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:7] Set("PJSIP/100-0000007f", "__TIMESTR=20170904-112024") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:8] Set("PJSIP/100-0000007f", "__FROMEXTEN=100") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:9] Set("PJSIP/100-0000007f", "__MON_FMT=wav") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:10] NoOp("PJSIP/100-0000007f", "Recordings initialized") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:11] ExecIf("PJSIP/100-0000007f", "0?Set(ARG3=dontcare)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:12] Set("PJSIP/100-0000007f", "REC_POLICY_MODE_SAVE=") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:13] ExecIf("PJSIP/100-0000007f", "0?Set(REC_STATUS=NO)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:14] GotoIf("PJSIP/100-0000007f", "3?checkaction") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (sub-record-check,s,17)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@sub-record-check:17] GotoIf("PJSIP/100-0000007f", "1?sub-record-check,out,1") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (sub-record-check,out,1)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [out@sub-record-check:1] NoOp("PJSIP/100-0000007f", "Outbound Recording Check from 100 to 0872910755") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [out@sub-record-check:2] Set("PJSIP/100-0000007f", "RECMODE=dontcare") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [out@sub-record-check:3] ExecIf("PJSIP/100-0000007f", "1?Goto(routewins)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (sub-record-check,out,7)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [out@sub-record-check:7] Gosub("PJSIP/100-0000007f", "recordcheck,1(dontcare,out,0872910755)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/100-0000007f", "Starting recording check against dontcare") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("PJSIP/100-0000007f", "dontcare") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [recordcheck@sub-record-check:3] Return("PJSIP/100-0000007f", "") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [out@sub-record-check:8] Return("PJSIP/100-0000007f", "") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [0872910755@from-internal:3] ExecIf("PJSIP/100-0000007f", "0 ?Set(CDR(accountcode)=)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [0872910755@from-internal:4] Set("PJSIP/100-0000007f", "MOHCLASS=default") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [0872910755@from-internal:5] Set("PJSIP/100-0000007f", "_NODEST=") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [0872910755@from-internal:6] Macro("PJSIP/100-0000007f", "dialout-trunk,3,0872910755,,off") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:1] Set("PJSIP/100-0000007f", "DIAL_TRUNK=3") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:2] GosubIf("PJSIP/100-0000007f", "0?sub-pincheck,s,1()") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:3] GotoIf("PJSIP/100-0000007f", "0?disabletrunk,1") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:4] Set("PJSIP/100-0000007f", "DIAL_NUMBER=0872910755") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:5] Set("PJSIP/100-0000007f", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:6] Set("PJSIP/100-0000007f", "OUTBOUND_GROUP=OUT_3") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:7] GotoIf("PJSIP/100-0000007f", "1?nomax") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (macro-dialout-trunk,s,9)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:9] GotoIf("PJSIP/100-0000007f", "0?skipoutcid") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:10] Set("PJSIP/100-0000007f", "DIAL_TRUNK_OPTIONS=T") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:11] Macro("PJSIP/100-0000007f", "outbound-callerid,3") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:1] ExecIf("PJSIP/100-0000007f", "0?Set(CALLERPRES(name-pres)=)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:2] ExecIf("PJSIP/100-0000007f", "0?Set(CALLERPRES(num-pres)=)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:3] ExecIf("PJSIP/100-0000007f", "0?Set(REALCALLERIDNUM=100)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:4] GotoIf("PJSIP/100-0000007f", "1?normcid") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (macro-outbound-callerid,s,7)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:7] Set("PJSIP/100-0000007f", "USEROUTCID=") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:8] Set("PJSIP/100-0000007f", "EMERGENCYCID=") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:9] Set("PJSIP/100-0000007f", "TRUNKOUTCID=111807") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:10] GotoIf("PJSIP/100-0000007f", "1?trunkcid") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (macro-outbound-callerid,s,15)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:15] ExecIf("PJSIP/100-0000007f", "1?Set(CALLERID(all)=111807)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:16] ExecIf("PJSIP/100-0000007f", "0?Set(CALLERID(all)=)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:17] ExecIf("PJSIP/100-0000007f", "0?Set(CALLERID(all)=)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:18] ExecIf("PJSIP/100-0000007f", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:19] ExecIf("PJSIP/100-0000007f", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:20] Set("PJSIP/100-0000007f", "CDR(outbound_cnum)=111807") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-outbound-callerid:21] Set("PJSIP/100-0000007f", "CDR(outbound_cnam)=") in new stack
[2017-09-04 11:20:24] WARNING[2188] func_cdr.c: CDR requires a value (CDR(variable)=value)
)[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:12] GosubIf("PJSIP/100-0000007f", "0?sub-flp-3,s,1()") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:13] Set("PJSIP/100-0000007f", "OUTNUM=0872910755") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:14] Set("PJSIP/100-0000007f", "custom=PJSIP") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:15] ExecIf("PJSIP/100-0000007f", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:16] ExecIf("PJSIP/100-0000007f", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:17] Macro("PJSIP/100-0000007f", "dialout-trunk-predial-hook,") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("PJSIP/100-0000007f", "") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/100-0000007f", "0?bypass,1") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/100-0000007f", "1?Set(CONNECTEDLINE(num,i)=0872910755)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/100-0000007f", "1?Set(CONNECTEDLINE(name,i)=CID:111807)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/100-0000007f", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)111807)") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/100-0000007f", "0?customtrunk") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:23] Dial("PJSIP/100-0000007f", "PJSIP/0872910755@111807,300,T") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] app_dial.c: Called PJSIP/0872910755@111807
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:24] NoOp("PJSIP/100-0000007f", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 1") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-dialout-trunk:25] GotoIf("PJSIP/100-0000007f", "0?continue,1:s-CHANUNAVAIL,1") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("PJSIP/100-0000007f", "RC=1") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("PJSIP/100-0000007f", "1,1") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (macro-dialout-trunk,1,1)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [1@macro-dialout-trunk:1] Goto("PJSIP/100-0000007f", "s-INVALIDNMBR,1") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (macro-dialout-trunk,s-INVALIDNMBR,1)
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s-INVALIDNMBR@macro-dialout-trunk:1] NoOp("PJSIP/100-0000007f", "Dial failed due to trunk reporting Address Incomplete - giving up") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s-INVALIDNMBR@macro-dialout-trunk:2] Progress("PJSIP/100-0000007f", "") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] pbx.c: Executing [s-INVALIDNMBR@macro-dialout-trunk:3] Playback("PJSIP/100-0000007f", "ss-noservice,noanswer") in new stack
[2017-09-04 11:20:24] VERBOSE[4794][C-00000034] file.c: <PJSIP/100-0000007f> Playing 'ss-noservice.ulaw' (language 'en')
[2017-09-04 11:20:26] VERBOSE[4794][C-00000034] pbx.c: Executing [h@from-internal:1] Macro("PJSIP/100-0000007f", "hangupcall") in new stack
[2017-09-04 11:20:26] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("PJSIP/100-0000007f", "1?theend") in new stack
[2017-09-04 11:20:26] VERBOSE[4794][C-00000034] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2017-09-04 11:20:26] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("PJSIP/100-0000007f", "0?Set(CDR(recordingfile)=)") in new stack
[2017-09-04 11:20:26] VERBOSE[4794][C-00000034] pbx.c: Executing [s@macro-hangupcall:4] Hangup("PJSIP/100-0000007f", "") in new stack
[2017-09-04 11:20:26] VERBOSE[4794][C-00000034] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/100-0000007f' in macro 'hangupcall'
[2017-09-04 11:20:26] VERBOSE[4794][C-00000034] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/100-0000007f'

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