October 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

NEC DSX 34B
by Sean Devoy 16 Oct '14

16 Oct '14

I have been asked to check if NEC DSX VOIP 34B phones can be used with FS. I notice there are none in the Interop list. If there is a known issue, it might save me a lot of time pounding sand to find out. Thanks, Sean

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Latest 1.4 still fails at FS-6849. Does anyone else notice it?
by Nuno Reis 16 Oct '14

16 Oct '14

Hi guys. I've just upgrade to FS 1.4.12 (b33a86c) and I see that JIRA FS-6849 <https://jira.freeswitch.org/browse/FS-6849> still happens. Isn't there anyone able to fix the issue using only the feedback provided by me on that JIRA? Do you really thing git bisect is the only way to go here? I really hope not. Looking forward to here from you. Best Regards, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> On Thu, Sep 25, 2014 at 4:54 AM, Nuno Reis <nreis(a)wavecom.pt> wrote: > Using gdb I've figured out that in mod_python.c:281: > ... > result = PyEval_CallObjectWithKeywords(function, arg, (PyObject *) NULL); > Py_DECREF(function); > ... > > variable result is always null when a python script trying to use the > session run the first time right after freeswitch startup. > That will make it enter the else logic in line 291 in the same file ending > with a crash. > I'll still have to drill more to find out why this is happening now. > Anyone has a clue on what's going on? > > BR, > > -- > > *Nuno Miguel Reis* | *Unified Communication** Systems* > M. +351 913907481 | nreis(a)wavecom.pt > WAVECOM-Soluções Rádio, S.A. > Cacia Park | Rua do Progresso, Lote 15 > 3800-639 AVEIRO | Portugal > T. +351 309 700 225 | F. +351 234 919 191 > *GPS > <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> > | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* > > [image: Description: Description: WavecomSignature] > <http://www.wavecom.pt/pt/wavecom/premios.php> > > [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> > > > > On Tue, Sep 23, 2014 at 3:53 AM, Nuno Reis <nreis(a)wavecom.pt> wrote: > >> This is the output after calling the script right after start: >> >> 2014-09-23 03:49:45.801129 [NOTICE] mod_python.c:212 Invoking py module: >> teste >> 2014-09-23 03:49:45.801129 [NOTICE] switch_cpp.cpp:1328 Testing123... >> 2014-09-23 03:49:45.801129 [NOTICE] switch_cpp.cpp:1328 Dumping current >> CDR INFO >> 2014-09-23 03:49:45.801129 [ERR] mod_python.c:293 Error calling python >> script >> 2014-09-23 03:49:45.801129 [ERR] mod_python.c:164 Python Error by calling >> script "teste": <type 'exceptions.AttributeError'> >> Message: 'PySwigObject' object has no attribute 'getXMLCDR' >> Exception: None >> >> Traceback (most recent call last) >> File: "/opt/freeswitch/scripts/teste.py", line 12, in handler >> >> All the following script calls work just fine. >> >> *Note:* If you want to test the code snippet i've added before don't >> forget to add: >> >> import freeswitch >> >> Looking forward to hear from you. >> >> Best Regards, >> >> -- >> >> *Nuno Miguel Reis* | *Unified Communication** Systems* >> M. +351 913907481 | nreis(a)wavecom.pt >> WAVECOM-Soluções Rádio, S.A. >> Cacia Park | Rua do Progresso, Lote 15 >> 3800-639 AVEIRO | Portugal >> T. +351 309 700 225 | F. +351 234 919 191 >> *GPS >> <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> >> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* >> >> [image: Description: Description: WavecomSignature] >> <http://www.wavecom.pt/pt/wavecom/premios.php> >> >> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> >> >> >> >> On Tue, Sep 23, 2014 at 3:44 AM, Nuno Reis <nreis(a)wavecom.pt> wrote: >> >>> Hi guys. >>> >>> I've reported this before and i can tell that it is still happening in >>> the latest v1.4. >>> >>> How reproduce: >>> >>> >>> 1. just create a dialplan that calls a python script >>> 2. dial that dialplan entry >>> >>> >>> Example code (works with anything that invokes *session*): >>> >>> #!/usr/bin/env python >>> # -*- coding: utf-8 -*- >>> >>> from __future__ import absolute_import >>> # http://docs.python.org/whatsnew/pep-328.html >>> >>> def handler(session, args): >>> freeswitch.consoleLog("notice", "Testing123...") >>> freeswitch.consoleLog("notice", "Dumping current CDR INFO") >>> freeswitch.consoleLog("info", str(session.getXMLCDR())) >>> freeswitch.consoleLog("notice", "The END!") >>> >>> >>> mod_pyhton crashes everytime the script is called right after a >>> freeswitch start (the very first time session is invoked inside a >>> mod_python python script). >>> After the first time (error), everything starts to work just fine. >>> This doesn't happen in e6b8b3e4f2deac73547bc9b9a2cb3e1d3147d101 from >>> v1.2.stable >>> >>> Should i open a jira on this? >>> >>> -- >>> >>> *Nuno Miguel Reis* | *Unified Communication** Systems* >>> M. +351 913907481 | nreis(a)wavecom.pt >>> WAVECOM-Soluções Rádio, S.A. >>> Cacia Park | Rua do Progresso, Lote 15 >>> 3800-639 AVEIRO | Portugal >>> T. +351 309 700 225 | F. +351 234 919 191 >>> *GPS >>> <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> >>> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* >>> >>> [image: Description: Description: WavecomSignature] >>> <http://www.wavecom.pt/pt/wavecom/premios.php> >>> >>> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> >>> >>> >>> >> >

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Asterisk 11.20 FreeSWITCH version: 1.5.15b
by Chris Allison 16 Oct '14

16 Oct '14

I have been at this for days now, I cant figure our why the FreeSWITCH-> Asterisk SIP registration is failing as the passwords look ok. Any help is much appreciated. Asterisk(10.237.192.53) sip.conf [gs-sbc1] username=gsvoice01 type=friend insecure=port,invite secret=ipscape@2014 qualify=yes host=10.237.192.68 ;192.168.202.60 dtmfmode=rfc2833 disallow=all canreinvite=no allow=alaw FreeSWITCH(10.237.192.68) sip internal profile <include> <gateway name="ips_voice"> <param name="username" value="gsvoice01"/> <param name="realm" value="10.237.192.53"/> <param name="from-domain" value="10.237.192.68"/> <param name="password" value="ipscape@2014"/> <param name="expire-seconds" value="60"/> <param name="retry-seconds" value="30"/> <param name="ping" value="25"/> </gateway> </include> FreeSWITCH console error 2014-10-16 00:07:46.105383 [ERR] sofia_reg.c:2312 ips_voice Registration Failed with status Forbidden [403]. failure #1 freeswitch@internal> 2014-10-16 00:07:47.105382 [WARNING] sofia_reg.c:502 ips_voice Failed Registration [403], setting retry to 30 seconds. Asterisk console error - debug <--- SIP read from UDP:10.237.192.68:5060 ---> REGISTER sip:10.237.192.53;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.237.192.68;rport;branch=z9hG4bK87tBt612BpgNp Max-Forwards: 70 From: <sip:gsvoice01@10.237.192.68>;tag=5jt9mZ5cD93pr To: <sip:gsvoice01@10.237.192.53> Call-ID: 747dfaba-54c8-11e4-bc60-81d0d9943fdf CSeq: 66373227 REGISTER Contact: <sip:10.237.192.68> Expires: 0 User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141008T204520Z~63734bcde0~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.237.192.68:5060 (no NAT) Sending to 10.237.192.68:5060 (no NAT) <--- Transmitting (no NAT) to 10.237.192.68:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.237.192.68;branch=z9hG4bK87tBt612BpgNp;received=10.237.192.68;rport=5060 From: <sip:gsvoice01@10.237.192.68>;tag=5jt9mZ5cD93pr To: <sip:gsvoice01@10.237.192.53>;tag=as0b2496ec Call-ID: 747dfaba-54c8-11e4-bc60-81d0d9943fdf CSeq: 66373227 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56dafaed" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '747dfaba-54c8-11e4-bc60-81d0d9943fdf' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.237.192.68:5060 ---> REGISTER sip:10.237.192.53;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.237.192.68;rport;branch=z9hG4bK9gm4U1j68y67H Max-Forwards: 70 From: <sip:gsvoice01@10.237.192.68>;tag=6UK2ptpgajt9K To: <sip:gsvoice01@10.237.192.68> Call-ID: aac7e260-54c9-11e4-bc61-81d0d9943fdf CSeq: 66373229 REGISTER Contact: <sip:gw+ips_voice@10.237.192.68:5060;transport=udp;gw=ips_voice> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141008T204520Z~63734bcde0~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.237.192.68:5060 (no NAT) Sending to 10.237.192.68:5060 (no NAT) <--- Transmitting (no NAT) to 10.237.192.68:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.237.192.68;branch=z9hG4bK9gm4U1j68y67H;received=10.237.192.68;rport=5060 From: <sip:gsvoice01@10.237.192.68>;tag=6UK2ptpgajt9K To: <sip:gsvoice01@10.237.192.68>;tag=as5eb1b731 Call-ID: aac7e260-54c9-11e4-bc61-81d0d9943fdf CSeq: 66373229 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53a1bd9f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'aac7e260-54c9-11e4-bc61-81d0d9943fdf' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.237.192.68:5060 ---> REGISTER sip:10.237.192.53;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.237.192.68;rport;branch=z9hG4bKatDXXv3957vtD Max-Forwards: 70 From: <sip:gsvoice01@10.237.192.68>;tag=6UK2ptpgajt9K To: <sip:gsvoice01@10.237.192.68> Call-ID: aac7e260-54c9-11e4-bc61-81d0d9943fdf CSeq: 66373230 REGISTER Contact: <sip:gw+ips_voice@10.237.192.68:5060;transport=udp;gw=ips_voice> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141008T204520Z~63734bcde0~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Authorization: Digest username="gsvoice01", realm="asterisk", nonce="53a1bd9f", algorithm=MD5, uri="sip:10.237.192.53;transport=udp", response="4057b65188f937c3a8cdbef65e3d8416" Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 10.237.192.68:5060 (no NAT) <--- Transmitting (no NAT) to 10.237.192.68:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.237.192.68;branch=z9hG4bKatDXXv3957vtD;received=10.237.192.68;rport=5060 From: <sip:gsvoice01@10.237.192.68>;tag=6UK2ptpgajt9K To: <sip:gsvoice01@10.237.192.68>;tag=as5eb1b731 Call-ID: aac7e260-54c9-11e4-bc61-81d0d9943fdf CSeq: 66373230 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Oct 16 00:16:26] NOTICE[25991]: chan_sip.c:28059 handle_request_register: Registration from '<sip:gsvoice01@10.237.192.68>' failed for ' 10.237.192.68:5060' - Wrong password Scheduling destruction of SIP dialog 'aac7e260-54c9-11e4-bc61-81d0d9943fdf' in 32000 ms (Method: REGISTER) ip-10-237-192-53*CLI>

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Time of day routing / too many stacked extensions
by Gavin Hamill 16 Oct '14

16 Oct '14

Hi all, I'm using FS 1.4.12 on CentOS 6.5 and I think I'm running into an issue with 'dialplan parsed all at once' when trying to implement the example based on https://wiki.freeswitch.org/wiki/Time_of_Day_Routing#Example_for_office_ope… : When I dial our main '9600' external DID, the call is answered and routed exactly as I want, however, there is a lot of ugly logging going on and I suspect that the desired behaviour is by chance rather than design. I'd love to hear some feedback on how to best implement the logic of 'Office open Mon-Fri 8am-6pm': 2014-10-16 08:49:48.515467 [NOTICE] switch_channel.c:1055 New Channel sofia/external/nobody(a)87.238.73.162 [ee5fb654-31c0-4c89-977e-b85ad4a0cb23] 2014-10-16 08:49:48.515467 [NOTICE] switch_core_session.c:2987 Execute execute_extension(441614249600_${office_status}) 2014-10-16 08:49:48.515467 [NOTICE] switch_core_session.c:2987 Execute execute_extension(441614249600_${office_status}) [.. 17 times in total...] 2014-10-16 08:49:48.515467 [NOTICE] switch_core_session.c:2987 Execute execute_extension(441614249600_${office_status}) 2014-10-16 08:49:48.515467 [ERR] switch_core_session.c:2918 Error sofia/external/nobody(a)87.238.73.162 too many stacked extensions 2014-10-16 08:49:48.515467 [NOTICE] switch_core_session.c:2987 Execute set(domain_name=freeswitch.bashton.eu) 2014-10-16 08:49:48.515467 [NOTICE] switch_core_session.c:2987 Execute set(effective_caller_id_name=${cidlookup(${caller_id_number})}) 2014-10-16 08:49:48.515467 [NOTICE] switch_core_session.c:2987 Execute ivr(bashton_9600_open) 2014-10-16 08:49:48.515467 [NOTICE] mod_sofia.c:2243 Pre-Answer sofia/external/nobody(a)87.238.73.162! <extension name="are_we_open" continue="true"> <condition wday="2-6" hour="8-18" break="never"> <action application="set" data="office_status=open" inline="true"/> <anti-action application="set" data="office_status=closed" inline="true"/> </condition> </extension> <extension name="DDI_9600" continue="true"> <condition field="destination_number" expression="441614249600"> <action application="execute_extension" data="441614249600_${office_status}"/> </condition> </extension> <extension name="DDI_9600_open"> <condition field="destination_number" expression="441614249600_open"> <action application="set" data="domain_name=$${domain}"/> <action application="set" data="effective_caller_id_name=${cidlookup(${caller_id_number})}"/> <action application="ivr" data="bashton_9600_open"/> </condition> </extension> <extension name="DDI_9600_closed"> <condition field="destination_number" expression="441614249600_closed"> <action application="set" data="domain_name=$${domain}"/> <action application="set" data="effective_caller_id_name=${cidlookup(${caller_id_number})}"/> <action application="ivr" data="bashton_9600_closed"/> </condition> </extension> Suggestions would be most welcome! :) Cheers, Gavin.

2 3

'Received DISCONNECT' numerous times before connecting
by Sudhanshu 16 Oct '14

16 Oct '14

Hi all, I am using a GSM Box with 15 sims to imitate a PRI card. But most of the time, just after the destination number is picked the call disconnects, x-lite shows an error 'Failed to establish call'. In around 1/6 times the call connects properly. Similar setups with PRI cards are working flawlessly. The relevant traces are as following: Sangoma stack trace: 2014-10-15 19:19:07.506516 [DEBUG] ftmod_sangoma_isdn_trace.c:224 [SNGISDN Q931] wp1 FRAME INCOMING: Prot Disc:Q.931/I.451 (0x08) Call Ref:0FC7 (Destination side) Type:DISCONNECT (0x45) Cause:coding:ITU-T(0) location:User(0) val:Normal call clearing(16) Progress Indicator:coding:ITU-T(0) location:Private network, local user(1) descr:In-band data ready(8) [ 08 02 8f c7 45 08 02 80 90 1e 02 81 88 ] Freeswitch debug info: 2014-10-15 19:19:07.506516 [INFO] ftmod_sangoma_isdn_stack_rcv.c:232 [s1c1][1:1] Received DISCONNECT (suId:1 suInstId:783 spInstId:863) Any suggestions? -- Sudhanshu

1 1

OpenSSL security advisory affecting FreeSWITCH users.
by Michael Jerris 16 Oct '14

16 Oct '14

Please be aware of the following OpenSSL advisory from today. The FreeSWITCH team recommends all users upgrade to openssl 1.0.1j as soon as possible: "A flaw in the DTLS SRTP extension parsing code allows an attacker, who sends a carefully crafted handshake message, to cause OpenSSL to fail to free up to 64k of memory causing a memory leak. This could be exploited in a Denial Of Service attack. This issue affects OpenSSL 1.0.1 server implementations for both SSL/TLS and DTLS regardless of whether SRTP is used or configured. Implementations of OpenSSL that have been compiled with OPENSSL_NO_SRTP defined are not affected.” https://www.openssl.org/news/secadv_20141015.txt

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one remote phone unable to make external calls
by Charles Bujold 15 Oct '14

15 Oct '14

Having an issue with a remote office phone being unable to make external calls. Any assistance on how to fix would be appreciated. The error I see in the log file are as follows: 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/15061239182 [BREAK] 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/15061239182 [BREAK] 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/15061239182 [BREAK] 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] sofia.c:6423 Channel sofia/external/15061239182 entering state [terminated][408] 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [NOTICE] sofia.c:7306 Hangup sofia/external/15061239182 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE] 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_channel.c:3222 Send signal sofia/external/15061239182 [KILL] 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/15061239182 [BREAK] 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_state_machine.c:472 (sofia/external/15061239182) Running State Change CS_HANGUP 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_state_machine.c:735 (sofia/external/15061239182) Callstate Change DOWN -> HANGUP 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_state_machine.c:737 (sofia/external/15061239182) State HANGUP 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] mod_sofia.c:413 Channel sofia/external/15061239182 hanging up, cause: RECOVERY_ON_TIMER_EXPIRE 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_state_machine.c:60 sofia/external/15061239182 Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_state_machine.c:737 (sofia/external/15061239182) State HANGUP going to sleep 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_state_machine.c:504 (sofia/external/15061239182) State Change CS_HANGUP -> CS_REPORTING 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/15061239182 [BREAK] 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_state_machine.c:472 (sofia/external/15061239182) Running State Change CS_REPORTING 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_state_machine.c:823 (sofia/external/15061239182) State REPORTING 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_state_machine.c:104 sofia/external/15061239182 Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_state_machine.c:823 (sofia/external/15061239182) State REPORTING going to sleep 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_state_machine.c:498 (sofia/external/15061239182) State Change CS_REPORTING -> CS_DESTROY 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/15061239182 [BREAK] 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.252830 [DEBUG] switch_core_session.c:1615 Session 81 (sofia/external/15061239182) Locked, Waiting on external entities 659bd7ed-f8e8-4224-81d2-cf1d4669c7c6 2014-10-15 13:52:07.272796 [DEBUG] switch_ivr_originate.c:3695 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE] 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.272796 [NOTICE] switch_core_session.c:1633 Session 81 (sofia/external/15061239182) Ended 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.272796 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/15061239182 [CS_DESTROY] 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.272796 [DEBUG] switch_core_state_machine.c:626 (sofia/external/15061239182) Running State Change CS_DESTROY 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.272796 [DEBUG] switch_core_state_machine.c:636 (sofia/external/15061239182) State DESTROY 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.272796 [DEBUG] mod_sofia.c:323 sofia/external/15061239182 SOFIA DESTROY 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.272796 [DEBUG] switch_core_state_machine.c:111 sofia/external/15061239182 Standard DESTROY 34d6aa01-f403-400a-8f53-9f06d25ea74b 2014-10-15 13:52:07.272796 [DEBUG] switch_core_state_machine.c:636 (sofia/external/15061239182) State DESTROY going to sleep 659bd7ed-f8e8-4224-81d2-cf1d4669c7c6 2014-10-15 13:52:07.272796 [INFO] mod_dptools.c:3234 Originate Failed. Cause: RECOVERY_ON_TIMER_EXPIRE 659bd7ed-f8e8-4224-81d2-cf1d4669c7c6 2014-10-15 13:52:07.272796 [NOTICE] switch_channel.c:4681 Hangup sofia/internal/426(a)192.168.222.153 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] 659bd7ed-f8e8-4224-81d2-cf1d4669c7c6 2014-10-15 13:52:07.272796 [DEBUG] switch_channel.c:3222 Send signal sofia/internal/426(a)192.168.222.153 [KILL] 659bd7ed-f8e8-4224-81d2-cf1d4669c7c6 2014-10-15 13:52:07.272796 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/426(a)192.168.222.153 [BREAK] 659bd7ed-f8e8-4224-81d2-cf1d4669c7c6 2014-10-15 13:52:07.272796 [DEBUG] switch_core_session.c:2888 sofia/internal/426(a)192.168.222.153 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) Thanks cjb

2 1

Lua Dialplan Application to turn down FS
by Bernhard Wendel 15 Oct '14

15 Oct '14

Hi, I'm using the following Lua-Script to check whether a conference exists and only let a caller join if so. The script is called via the following dialplan /<action application="lua" data="/path/to/script.lua $1" />///, where $1 is the destination number. 1. -- Connect to a database 2. local dbh = freeswitch.Dbh("pgsql://hostaddr=127.0.0.1 dbname=database user=username password='password' application_name='my-lua-script'") 3. assert(dbh:connected()); 4. -- Execute SQL 5. dbh:query(string.format("SELECT COUNT(*) as \"MyCount\" FROM \"MyTable\" WHERE ColA = '%s'", argv[1]), function(row) 6. --freeswitch.consoleLog("INFO", "Results: " .. row.MyCount) 7. 8. dbh:release() 9. 10. if (row.MyCount== '1') then 11. freeswitch.consoleLog("INFO", "User is joining conference at " .. argv[1]) 12. 13. session:answer() 14. session:execute("conference", argv[1] .. "@wideband") 15. end 16. session:hangup() 17. end) Line 8 instructs FS to return the postgres-Connection back to the pool - which is as expected. When a user now leaves the conference, FS throws this errors: 2014-10-12 10:32:30.065924 [DEBUG] switch_cpp.cpp:694 CoreSession::hangup 2014-10-12 10:32:30.065924 [CRIT] switch_pgsql.c:368 An error occurred trying to consume input for query (200): connection pointer is NULL 2014-10-12 10:32:30.065924 [CRIT] switch_pgsql.c:122 No DB Connection 2014-10-12 10:32:30.065924 [ERR] switch_core_sqldb.c:1177 ERR: [SELECT COUNT(*) as "MyCount" FROM "MyTable" WHERE ColA = 'conf-1234'][connection pointer is NULL] 2014-10-12 10:32:30.065924 [ERR] freeswitch_lua.cpp:446 DBH NOT Connected. 2014-10-12 10:32:30.065924 [DEBUG] switch_cpp.cpp:1075 sofia/internal/Alice(a)my.pbx.com destroy/unlink session from object Now, I could live with that. BUT: The problem is, that sometimes, when a user leaves the conference, FS crashes. I've narrowed down the main cause to the above lua script. If I comment line 8 out, FS stops crashing. This will also stop FS from reusing Database-Connections. Now my questions: 1. I don't use `dbh` after line 8. So where does the error come from? 2. How can I stop the error? 3. How can I rewrite the lua script to have both a) non-crashing FS and b) reusing of connection? Regards, Bernhard

3 6

Freeswitch - Create users using add_user.pl
by Guilherme Dantas 15 Oct '14

15 Oct '14

Hello, I would like to know how can I create groups in the freeswitch. I already created new users and extensions changing the dialplan and using the add_user.pl script. How could I determinate a group by a number? Do I have to change the dialplan? Thanks in advance, <http://www.inovax.com.br/> Description: Description: www.inovax.com.br Guilherme Vieira Dantas - Estagiário <mailto:guilherme.dantas@inovax.com.br> guilherme.dantas(a)inovax.com.br Skype: inovax5554 _____ Rua Paulo Emídio Barbosa, 485, Quadra 1, Centro de Excelência TIC Ilha da Cidade Universitária Rio de Janeiro RJ - CEP 21941-907

2 1

Pickup on Ringgroup
by Pedro Santos 15 Oct '14

15 Oct '14

HI, i have a bridge to a ringgroup like this bridge {continue_on_fail=true,ignore_early_media=true,originate_continue_on_timeout=true,leg_timeout=20,ringback=$${bong-ring}}sofia/internal/400%voip.spsul.local|sofia/internal/401%voip. spsul.local|sofia/internal/448%voip.spsul.local I want to use group-intercept or intercept-ext but it does'n work. Any ideas ?

2 1