@asifahmed009 wrote:
Hi Guys,
One touch monitor works fine.Means middle of conversations, activate the call recordings is working.
Is it possible to end recording redial *1 or define any other feature code for disconnecting recording.
Any help would be appreciated.
Posts: 1
Participants: 1
@cholegmw wrote:
Hi,
I have configured SIP trunk with registration.
Everything was working fine till some other configuration changed and configuration apply (sip reload). Then outgoing registration got status "Unregistered" and I can make calls. When I run "amportal restart" or system reboot, SIP registration come back to status "Registered" and everything works fine.
SIP trunk was configured with my provider thru L3VPN tunnel, and because that there is no NAT between.I was just make a new instalation of AsteriskNow 6.12.65 (64-bit) and configure from begining, adn I have same problem.
Registration string is: PBX011DXXXX000@ims.provider.com:PASSWORD:PBX011DXXXX000@ims.provider.com:5060/PBX011DXXXX000
Asterisk ver. 11.21.2
Chan_Sip Registry
Host dnsmgr Username Refresh State Reg.Time
109.245.XXX.XXX:5060 N PBXXXXXXX 120 Unregistered
1 SIP registrations.On manual sip reload (module relaod chan_sip.so):
WARNING sip_parse_register_line: Format for registration is [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] at line 10WARNING35988: sched.c:489 sched_settime: Bug likely: Negative time interval -34284 (interpreted as 4294933012 ms) requested!
Thanks in advance!
Mladen
Posts: 1
Participants: 1
@dcitelecom wrote:
I uninstalled the digium phones module, rest api, and rest apps and even though I selected remove from disk I have the 3 directories below on the disk. Is t safe to remove them?
restapiand digium_phones is empty. restapps is full of zip files./var/www/html/restapi
/var/www/html/restapps
/var/www/html/digium_phones
Posts: 1
Participants: 1
@RKM wrote:
I have installed FreePBX 14.0.1alpha34.
Prior to doing that, I used the Backup & Restore module, to backup my FreePBX 13.0.188.8 configuration.
When I attempted to restore that backup to 14.0.1alpha34, it gives me an error that the 'calendar_id' is missing.
Now I'm stuck with the 'Apply Config' button in a permanent status where I cannot apply this change (or any future changes of any sort) and I cannot revert back what I've done.
In short, the system appears to be hosed. Is there any way to make v14 more forgiving when importing backups from prior versions?
Thank you in advance.
For reference, this is the full copy of the error message:
exit: 1 PDOException: SQLSTATE[42S22]: Column not found: 1054 Unknown column 'calendar_id' in 'field list' in file /var/www/html/admin/modules/findmefollow/Findmefollow.class.php on line 714 Stack trace: 1. PDOException->() /var/www/html/admin/modules/findmefollow/Findmefollow.class.php:714 2. PDOStatement->execute() /var/www/html/admin/modules/findmefollow/Findmefollow.class.php:714 3. FreePBX\modules\Findmefollow->get() /var/www/html/admin/modules/findmefollow/functions.inc.php:379 4. findmefollow_get() /var/www/html/admin/modules/findmefollow/functions.inc.php:1031 5. findmefollow_getdestinfo() /var/www/html/admin/libraries/usage_registry.functions.php:300 6. framework_identify_destinations() /var/www/html/admin/libraries/usage_registry.functions.php:351 7. framework_list_problem_destinations() /var/lib/asterisk/bin/retrieve_conf:817
Posts: 6
Participants: 3
@OLO wrote:
installed freepbx on centos 7 of AWS, I can go to admin webpage via public ip, then add extension, however the extension can't connect server(public ip), also add security group for both TCP and UDP on port 5060,
anything wrong?
Posts: 6
Participants: 2
@Driver wrote:
Hello,
On an older FreePBX version I was able to create custom extensions from the script and reload sip. Now the custom extensions are also visible in 'sip show peers', they are also qualified but it is impossible to make a call to them.I've got:
Playback("SIP/114-00000005", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer")In sip debug there is a
SIP/2.0 503 Service Unavailablemessage orSIP/2.0 405 Method Not Allowed.When I make 'sip reload' there are many warnings like this:
chan_sip.c:31471 build_peer: Invalid session-timers '' at line xxx of sip.confSettings for built-in and custom extensions are the same.
What can I do in this situation?
Posts: 3
Participants: 2
@dcitelecom wrote:
First time I try to do a restore on another server of SQL table backed up with FreePBX distro and it fails with permission denied. Any idea what the error is?
Starting restore.
Initializing Restore...
Running pre-restore hooks, if any...
Restoring files (this may take some time)...
/bin/tar: ./mysql-7.sql: Cannot open: Permission denied
/bin/tar: Exiting with failure status due to previous errors
File restore complete!
Running post-restore hooks, if any...
Cleaning up...
Restore complete!
Reloading...
Done!
Posts: 2
Participants: 1
@tigger1197 wrote:
I just installed several commercial apps. Now when I apply conifg it fails and gives error 255 on restapps.
exit: 255
Unable to continue. SQLSTATE[HY000]: General error: 130 Incorrect file format 'restapps_settings':: in /var/www/html/admin/libraries/utility.functions.php on line 2050 /var/www/html/admin/modules/restapps/Restapps.class.php(63): die_freepbx('SQLSTATE[HY000]...')
1 /var/www/html/admin/modules/restapps/Restapps.class.php(1144): FreePBX\modules\Restapps->getControllerSettings()
2 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(105): FreePBX\modules\Restapps->doDialplanHook(Object(extensions), 'asterisk', 900)
3 /var/lib/asterisk/bin/retrieve_conf(866): FreePBX\DialplanHooks->processHooks('asterisk', Array)
4 {main}
Posts: 3
Participants: 2
@dcitelecom wrote:
FreePBX 13.0.190.7 , Asterisk 13
I recently enabled the SIP jitter buffer and we are now getting the Warnings below with every call. Is this a bug or is something wrong with my setup? If I turn off jitter buffer, the warnings stop.
[2016-11-24 15:18:03] WARNING[14452][C-0000138f]: chan_iax2.c:1239 jb_warning_output: Resyncing the jb. last_delay 0, this delay -7594, threshold 500, new offset 7594
[2016-11-24 15:18:03] WARNING[15653][C-0000138f]: chan_iax2.c:1239 jb_warning_output: Resyncing the jb. last_delay 0, this delay -2134237, threshold 500, new offset 2134237
[2016-11-24 15:18:50] WARNING[20817][C-00001390]: chan_iax2.c:1239 jb_warning_output: Resyncing the jb. last_delay 0, this delay -8803, threshold 500, new offset 8803
Posts: 1
Participants: 1
@Issue wrote:
Hi,
Is it possible in FreePBX to Play Music on Hold (or announcements) after a delay, like if someone doesn't pick up the Phone after 20 seconds of ringing FreePBX starts an announcement like "Sorry it's taking longer than usual...".
Posts: 1
Participants: 1
@mcisar wrote:
I know there are several places within FreePBX where the system does not accept compounded recordings but I don't recall ever seeing the reason why posted anywhere. So now that I'm working on my turkey-midnite snack... can anybody post a one-liner explanation of why compound recordings work in some places but not others?
With that pesky question out of the way... is there an easy way to merge multiple audio files into one if I want to build a custom non-compounded phrase out of the stock recordings?
Cheers,
Mike
Posts: 2
Participants: 2
@mcisar wrote:
Just wondering if any of the developers can comment on the feasibility of having a "Ring Group & Page" functionality, similar in concept to "Park & Page"; whereby an announcement could be paged when a call is transferred to a ring group.
I can see this equally being applied to a queue (page upon call entry to queue), and actually to just a plain extension by itself (page when a call is directed to an extension).
I can see this as being very useful in some of the small 2 and 3 person offices that I deal with where they still prefer to answer the phone immediately rather than dropping to voicemail, but they may be frequently away from their desk doing other tasks. You would then have a scenario where you'd could have "sales group, incoming call" (ring group) or "warehouse phone is ringing" (direct extension).
Can anyone else see scenarios where this might be useful?
If it seems feasible I'll go and submit a feature request.
Cheers,
Mike
Posts: 1
Participants: 1
@avayax wrote:
What is the standard Opus codec bitrate on the new beta release of SNG7, Asterisk 13, and is it adjustable?
By the way, I am trying to make Opus work on SNG7 and Yealink T23G phones (latest V81 firmware supports Opus), but no luck.
Calling another phone as well as into an internal conference returns "SIP/2.0 488 Not acceptable here" and "process_sdp: No compatible codecs, not accepting this offer!".
Posts: 1
Participants: 1
@warheat1990 wrote:
Hello,
I subscribe to ISP that gave me internet, cable TV, and phone service. ISP gave me a fiber optic modem router combo which came with RJ45 port and RJ11 port.
I plugged in my PC/STB into the RJ45 port for internet and cable TV. I also plug in my analog phone into the RJ11 port. Basically everything is plug and play and no settings required on my end.
Today I look around at the modem web interface and found that they have SIP account information inside it.
So my question is can I use this SIP account and use it on my FreePBX device? I already tried the account in X-Lite but it's not working.
![]()
Keep in mind that I'm absolutely clueless about VOIP and SIP stuff so any help will be appreciated.
Posts: 1
Participants: 1
@Marbled wrote:
Hi!
I noticed yesterday that a call to one of my DAHDI extension which doesn't have voicemail doesn't show up in CDR...
Is this
- a misconfiguration on my side?
- by design?
- a bug?
Any idea?
Thank you and have a nice day!
Nick
Posts: 1
Participants: 1
@friday1970 wrote:
So, my fellow network admin and I have painted ourselves into a little corner. Years ago, we've set up a trixbox (v2.8.0.4), and for the most part, it still runs great. The phones are all Polycom and Snom, and everything is perfect
But...
This network we manage for a client has grown. Business has picked up, so more phones and PCs have been added. Now, we're running out of IPs on this class C network. The phones and PCs are all on the same subnet, and now we're probably down to about 5 free addresses left. IP conflicts are starting to be a problem when the users there take it upon themselves to add equipment.
So, I've have drawn up in my head to create another subnet, a 10.10.10.x/24 network, and make this the phone only network. So far, all I've done is added a subinterface to trixbox, a eth0:1 entry. It's pingable from any 10.10.10.x device. I've also started to create a 10.10.10.0 DHCP scope with mac reservations using the macs from the phones and will activate it once I think I have things figured out.But I do have a few questions:
First:
How do the phones communicate with the server and then to the outside world? Do the phones relay voice through the server directly, and then the server sends sip packets through our SIP trunk? Or do the phones receive a token from the server and then the phones send packets through the SIP trunk itself with the server monitoring it all? The trixbox server will still retain it's original IP address on the 10.18.157.x along with the new 10.10.10.1 secondary IP, as the provider's sip trunk LAN port is 10.18.157.x.. Also, is there any routing I would need to enable on the trixbox, and/or routing entries?Second. I read elsewhere that for trixbox, I would add entries into the sip_general_custom.conf file, a bindport = 5060 entry and a bindaddr = 0.0.0.0 entry. But, would I need to comment out the #sip_general_custom.conf entry in sip.conf and then reboot? Also, sip.conf doesn't have much in it, with no bindport or bindaddr entries. However, sip.conf.0 does contain bindport and bindaddr entres, showing both 5060 and 0.0.0.0. Does Trixbox currently use the sip.conf.0 as a running file, and would this mean it is already listening to any available IPs?
Sorry for so many questions for a first time poster. But, we want to have the legwork done and functioning before we commit to an after hours, probably all nighter of work, changing TFTP entries on every phone, checking, testing, etc.
Posts: 4
Participants: 3
@RKM wrote:
I previously had FreePBX 13 installed and running UCP with no issues.
In FreePBX 14.0.1beta1, when connecting to UCP, a "loading" modal shows and the web page then locks-up.
This is in latest Chrome and MSIE both.
I did some searching and I noticed there were issues in the past with UCP due to webRTC / issues with requiring the cert. All of my cert issues are fine on this system. I also completely uninstalled the UCP and all dependency modules, then re-installed, just to be safe.
I only receive two errors in Chrome Java Debug Console (tested on two different machines, same issues on both):
Uncaught ReferenceError: Cookies is not defined(…)
Error in event handler for (unknown): TypeError: Cannot read property 'shortcutLock' of undefined
at chrome-extension://oknpjjbmpnndlpmnhmekjpocelpnlfdi/js/content.js:32:32Is this just something obvious I haven't turned-up in my searches?
Posts: 2
Participants: 2
@gooddisk92 wrote:
Hi,
I am able to capture the Caller ID in SIP Debug but I have no knowledge how to extract it.
Therefore the pbx shows Anonymous from all incoming calls.The SIP provider require the PBX to use P-Asserted-Identity to display the Caller Line Identity (CLIP).
I have tried the context provided by freepbx such as [from-pstn-toheader] but it did not extract the CLIP from the Sip Header.
Please advise.
Equipment Version
FPBX-13.0.190.7(13.12.1)
Asterisk 13.12.1Testing scenario
+60399999999 calling +60388888888Outcome
DID (+60388888888) is detected correctly via PAI and sent to Extension
But incoming Calling Line Identity appears as Anonymous.The following is the output of the SIP debug
<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
INVITE tel:+60388888888 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK0byz189xyt118w8sxzy121x90;Role=3;Dpt=7862_36;TRC=ffffffff-ffffffff
Route:
Record-Route:
Call-ID: xj8plg1gipp0zlr2i8a2vijli88j2a38ATATS.rcatshw01.ims.sipprovider.com.151
From: ;tag=r1gijhja-CC-151
To:
From: ;tag=r1gijhja-CC-151
To:
CSeq: 1 INVITE
Alert-Info: info=pattern1
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER
Contact: <sip:+60399999999ATXX.XX.XX.XX:5060>
Max-Forwards: 62
Supported: 100rel,timer
Session-Expires: 1200
Min-SE: 1200
P-Called-Party-ID:
P-Access-Network-Info: xDSL;dsl-location="XXX_V1032 eth 0/11/0/13:400";"location-info=XXX_V1032 eth 0/11/0/13:400"
P-Early-Media: gated
Content-Length: 219
Content-Type: application/sdpv=0
o=HuaweiATS9900 5313141 5313141 IN IP4 XX.XX.XX.XX
s=Sip Call
c=IN IP4 XX.XX.XX.XX
t=0 0
m=audio 41170 RTP/AVP 8 0 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
<------------->
--- (20 headers 11 lines) ---
Sending to XX.XX.XX.XX:5060 (NAT)
Sending to XX.XX.XX.XX:5060 (NAT)
Using INVITE request as basis request - xj8plg1gipp0zlr2i8a2vijli88j2a38 ATATS.rcatshw01.ims.sipprovider.com.151
Found peer 'multisip' for 'Anonymous' from XX.XX.XX.XX:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port XX.XX.XX.XX:41170
Looking for +60388888888 in from-pstn (domain )
sip_route_dump: route/path hop:
Posts: 3
Participants: 3
@dcitelecom wrote:
I know Cyberlink is the official FreePBX hosting provider so I understand that the official community needs to endorse them but I need a local IP in Asia and Vultr has servers in Singapore whereas Cyberlink doesn't.
Vultr also offers more memory and additional CPU cores as options for the VMs so there is room to grow whereas with Cyberlink I'd have to switch to a virtual sever if I need more resources.
The system will have 35-50 simultaneous calls on FreePBX 10, Asterisk 13 with a2billing and g729 codecs for the calls. Any comments? Please share your experiences with both providers. Many thanks.
Posts: 3
Participants: 2
@batchen wrote:
Hi,
i have Asterisk 13.10.0 on SHMZ release 6.6 machine.
we have a problem that too many recordings file types - wav , ulaw , sln48ones in a while i see recordings are gatherd to 20G !
whe i hear the recordings its a real conversation between extentions and client and
the original recording is saved at the /var/spool/asterisk/monitori dont understand what make this recordings copy them selfs.
logs have no record on this.file name exsample :
temp.1480086466900.ulawthanks
Posts: 3
Participants: 2
