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call_timeout
by Federico Beffa 27 Aug '11

27 Aug '11

I've just traced a call on my Asterisk system and ... well, I get exactly the same behavior. The provider must have changed settings recently without any notice. In fact my mailbox is never reached anymore since I'm waiting for 30 seconds, but they CANCEL after 20... sorry for making you loose time... In any case I've learned how to trace calls in FreeSwitch :-) Regards, Fede

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call_timeout
by Federico Beffa 27 Aug '11

27 Aug '11

Is there a way to force the provider's proxy to wait longer with some kind of signal? I'm using "ring_ready" before transfer/bridge. Should I add or replace it with something else? On another system I have Asterisk installed (I'm planning to replace it with freeswitch) and for some reason the same provider does not drop the call for much longer. I guess I will have to trace the Asterisk call and compare... Thanks, Fede >Your provider seems to be dropping the call on you. > >see - > >recv 415 bytes from udp/[195.190.1xx.2xx]:5060 at 18:57:29.322712: > ------------------------------------------------------------------------ > CANCEL >sip:gw+ticinocom_private at 84.55.2xx.xx:5080;transport=udp;gw=ticinocom_private >SIP/2.0 > Via: SIP/2.0/UDP >195.190.1xx.2xx:5060;branch=z9hG4bK-d8754z-81e78d37b38d1735-1---d8754z-;rport > Max-Forwards: 70 > To: <sip:41916001220 at 195.190.1xx.2xx> > From: >"0765681626"<sip:0765681626 at 195.190.1xx.2xx>;tag=36agpbj7t724x3im.o > Call-ID: 7b9707bd-9e83a87-521dc9e3-3979 at 62.65.1xx.5x > CSeq: 737 CANCEL > Content-Length: 0

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call_timeout
by Federico Beffa 27 Aug '11

27 Aug '11

The call monitored in the siptrace was as follows: the call originated from my mobile phone. The destination was my desk phone, a Snom 300. The Snom 300 is registered with FreeSwitch. The call from the mobile phone arrives to FreeSwitch through my provider. After ca. 19 secs the mobile phone seems to hangup and start a new call (all automatically with no manual intervention). If instead of calling from my mobile phone, I originate the call from X-lite (softphone), then, after ca. 19 secs the call is terminated. I'm trying to figure out how to prevent the call to be terminated or re-started after these 19 seconds. Hope this is clearer. Regards, Fede >Can you re-phrase your first paragraph? > >On Friday, 26 August 2011, Federico Beffa <beffa at ieee.org> wrote: >> Dear Jeff and Michael, >> >> thanks for the answer. I've put siptrace and log on pastebin: >> >> http://pastebin.freeswitch.org/17205 >> >> The case I've monitored was me calling with my mobile phone my desk phone >(Snom 300). The desk phone is registered to Freeswitch and the call goes >through my provider. Note that after ca. 19 secs my mobile phone makes a new >call, while if I try to do the same, but calling with X-lite, then X-lite >just hangs up.

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call_timeout
by Federico Beffa 26 Aug '11

26 Aug '11

Dear Jeff and Michael, thanks for the answer. I've put siptrace and log on pastebin: http://pastebin.freeswitch.org/17205 The case I've monitored was me calling with my mobile phone my desk phone (Snom 300). The desk phone is registered to Freeswitch and the call goes through my provider. Note that after ca. 19 secs my mobile phone makes a new call, while if I try to do the same, but calling with X-lite, then X-lite just hangs up. The problem seems to be the CANCEL signal at line 459. Am I right in interpreting this signal as originating from my provider? If that is the case, then is there a way to tell my provider's proxy to wait longer? If my interpretation is correct, then I'm surprised that the same does not happen with Asterisk, where the call does not terminate (nor is re-started) for more than 1 minute. Thanks for any advise! Fede

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Error SRTP unprotect
by Aakash 26 Aug '11

26 Aug '11

Hi Guys, I am using latest git version.Sometimes i am getting an error as below [ERR] switch_rtp.c:2559 Error: SRTP unprotect failed with code 7 (auth check failed) The calls got disconnected when this error hits. Thanks, Aakash -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-SRTP-unprotect-tp673023… Sent from the freeswitch-users mailing list archive at Nabble.com.

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PRI to VoIP gateway devices
by Michael Collins 26 Aug '11

26 Aug '11

Hello gang, If anyone out there has a PRI to VoIP gateway device that we can borrow (or that you'd like to donate) to the FreeSWITCH project please let me know. We are looking for something like the Vegastream 400 or the Audiocodes Mediant 600. If you have anything like these that are just collecting dust on a shelf somewhere please contact me off list. Thanks! -Michael

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call_timeout
by Federico Beffa 26 Aug '11

26 Aug '11

Hi, I've setup the dialplan such that when I receive a call from a provider (public context), the call get transferred to an extension in the default context (in a similar way as the default config). However, no matter what I set for call_timeout, the call is always terminated after ca. 19 seconds with a cause ORIGINATOR_CANCEL. I would like to extend the timeout period. This is my public entry for the incoming call: <include> <extension name="incoming_call"> <condition field="destination_number" expression="^XXXXX$"> <action application="set" data="domain_name=$${domain}"/> <action application="export" data="call_timeout=60"/> <action application="export" data="bridge_answer_timeout=60"/> <action application="export" data="originate_timeout=60"/> <action application="ring_ready"/> <action application="bridge" data="user/1000@${domain_name}"/>  <action application="hangup"/> </condition> </extension> </include> And this is part of the log: 2011-08-25 23:57:30.920053 [NOTICE] sofia.c:5226 Ring-Ready sofia/external/0916001220! 2011-08-25 23:57:30.920053 [NOTICE] mod_sofia.c:2366 Ring-Ready sofia/internal/1001@xxxxx! 2011-08-25 23:57:30.920053 [NOTICE] switch_ivr_originate.c:481 Ring Ready sofia/internal/1001@xxxxx! 2011-08-25 23:57:30.960053 [NOTICE] sofia.c:5226 Ring-Ready sofia/internal/ sip:1000@192.168.0.5:31148! 2011-08-25 23:57:30.960053 [NOTICE] switch_ivr_originate.c:481 Ring Ready sofia/external/0916001220(a)195.190.166.249! 2011-08-25 23:57:50.900063 [NOTICE] sofia.c:5872 Hangup sofia/external/ 0916001220(a)195.190.166.249 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2011-08-25 23:57:50.900063 [NOTICE] switch_ivr_originate.c:3144 Hangup sofia/internal/sip:1000@192.168.0.5:31148 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2011-08-25 23:57:50.900063 [NOTICE] switch_ivr_originate.c:2436 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2011-08-25 23:57:50.900063 [INFO] mod_dptools.c:2696 Originate Failed. Cause: ORIGINATOR_CANCEL 2011-08-25 23:57:50.900063 [NOTICE] sofia.c:5872 Hangup sofia/external/0916001220 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1353 Session 11 (sofia/external/0916001220(a)195.190.166.249) Ended 2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1355 Close Channel sofia/external/0916001220(a)195.190.166.249 [CS_DESTROY] 2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1353 Session 12 (sofia/internal/sip:1000@192.168.0.5:31148) Ended 2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1355 Close Channel sofia/internal/sip:1000@192.168.0.5:31148 [CS_DESTROY] 2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1353 Session 10 (sofia/external/0916001220) Ended 2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1355 Close Channel sofia/external/0916001220 [CS_DESTROY] 2011-08-25 23:57:50.900063 [INFO] mod_dptools.c:2696 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2011-08-25 23:57:50.900063 [NOTICE] mod_dptools.c:2815 Hangup sofia/internal/1001@xxxxx [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1353 Session 9 (sofia/internal/1001@xxxxx) Ended 2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1355 Close Channel sofia/internal/1001@xxxxx [CS_DESTROY] I've tryied with "transfer" and with "bridge" also setting [leg_timeout=60], but no luck. I've changed all timeout related values in default.xml as well. If the call comes from a phone registered locally, then the timeout works fine. The problem only exists with calls coming from outside (the public context). However, the timeout does seems to be related to freeswitch and not the provider as I have another system running Asterisk and there the problem (using the same provider accounts) does not exist. I would really appreciate any suggestion on how to extend calls timeout. Regards, Fede

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Newbie - External Phone setup question
by Charles Bujold 26 Aug '11

26 Aug '11

Trying to get a phone located in another office registered to the Freeswitch server. I created an "external5090.xml" in the sip external folder as per the wiki and then set the phone to connect to 5090. But I cannot get the phone to register. When I look at wireshark I see a request from the phone to register and then I see an error "SIP:403 Forbidden" from the server to the Phone. I have created the extension in the Directory folder with the other internal extensions but I'm wondering if I'm missing something in the 1010extension.xml for it to work. (maybe the extension is located in the wrong folder, or some identification to say it is not in the local LAN?) I have checked the name and password and they are correct which seems to indicate the issue is at the server level. Any guidance on how to setup an external phone to Freeswitch would be appreciated. Thanks CJB

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Mod_Managed Mono is not following the system LD Path
by R H 26 Aug '11

26 Aug '11

Hey Everyone, I have recently been asked to set up a FreeSwitch server on a company development server and the development management wants to use Mono C# as our language for adding logic to the platform. Following the instructions at http://wiki.freeswitch.org/wiki/Mod_managed I have been able to successfully compile freeswitch on a Suse 11.4 box, with a custom installation of Mono 2.8 and I can *successfully execute Demo.csx*. But here is the problem, the only way I can get Demo.csx to run correctly without getting a NullReferenceException is to start the server from the freeswitch mod directory: EG. $> cd /usr/local/freeswitch/mod/ $> /usr/local/freeswitch/bin/freeswitch * [SUCCESS!, Demo.csx will run correctly]* $> cd / (or any other directory) $> /usr/local/freeswitch/bin/freeswitch [FAIL, The server starts, calls work, everything is fine EXCEPT Demo.csx now throws *NullReferenceExceptions when I hit it with a call that is routed to the script by the dial plan]* - I have tried changing the *dllmap* default to the mono installation, I have tried adding /usr/local/freeswitch/mod/*FreeSWITCH.Native.dll.config*with the following text inside: """ <configuration> <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/> </configuration> """ - I have tried adding /usr/local/freeswitch/mod/ to the system ld path (/etc/ld.so.conf). I also reloaded the ld cache (ldconfig) - I have tried copying FreeSWITCH.Managed.dll to the existing ld path (/usr/local/lib/FreeSWITCH.Managed.dll) - I even created a Console HelloWorld.exe that imports the Freeswitch libraries and ran it by simply executing: $> mono

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Making Freeswitch listen on 2 IP addresses
by Pete Kelly 26 Aug '11

26 Aug '11

Hi Is it possible to make freeswitch listen on more than one IP for requests within the same profile? I would like to assign a VIP to a pair of freeswitch boxes, for the purposes of failover if one of the boxes should fail or go down. The VIP can then be used by web applications which have to send XMLRPC requests to freeswitch. In addition I would like each of the boxes to listen on their "normal" eth0 IP too, so systems like opensips which support load balancing can evenly distribute load across the two boxes. My profile.xml file currently has this entry: <param name="sip-ip" value="$${local_ip_v4}"/> So I can't see how freeswitch can be told to listen on more than one IP for a profile. Is this possible?

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