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Compilation Error In Freeswitch
by Arpit Gadle 23 Apr '13

23 Apr '13

Dear All, I am getting this compilation error on one of my box oma_ss7_la-ftmod_sangoma_ss7_cfg.o src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c: In function 'ftmod_ss7_mtp2_dlsap_config': src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:878: error: expected expression before ';' token src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:879: error: expected expression before ';' token src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:880: error: expected expression before ';' token src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:881: error: expected expression before ';' token src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:882: error: expected expression before ';' token src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:883: error: expected expression before ';' token Thanks & Regards, Arpit Gadle Mobile : +91-9303964208

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calls dropped after refer with replaces
by Michel Brabants 23 Apr '13

23 Apr '13

Hello, when I send a refer with replaces from my polycom, the calls are dropped with which I'm trying to do an attended transfer. When leg arrives in CS_PARK and the other in CS_HIBERNATE. An unbridge happens afterwards on the leg that sent the refer-replaces, but in the end all legs are dropped. Any idea why this could be happening? I tried to look at the sourcode, but I couldn't find my problem yet. Thanks, Michel

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freeswitch not replying on port according to srv-record
by Michel Brabants 23 Apr '13

23 Apr '13

Hello, one more question. While performing tests, I noticed that freeswitch didn't respond to the port that is specified in the srv-record, which is in the contact-address. This is when it receives an incoming invite in one of the defined gateways in freeswitch. The srv-record has 5062 in it, but freeswitch doesn't seem to try to resolve the srv-record. Kind regards, Michel

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Catch events from ESL socket
by Anton Vojlenko 23 Apr '13

23 Apr '13

Hello, I am using skypopen module and i need catch API answers from skype client. I wrote simple perl script: require ESL; my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); while (1){ $con->sendRecv("log 7"); $e = $con->bgapi("skypopen sk1 PING"); #$uuid = $e->getHeader("Job-UUID"); #print $e->serialize("plain"); $n = 2; while($n){ $event = $con->recvEventTimed(100); if ($event and $event->getHeader("Log-File") eq "skypopen_protocol.c") { push(@dmesg, $event->getBody()); } $n--; } print "@dmesg\n"; sleep(3); @dmesg = (); } But sometimes I get empty array, because script catch differect events. How to catch events from ESL socket by UUID?

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Hunting down bugs with git bisect
by Michael Collins 23 Apr '13

23 Apr '13

Hello gang! I'm soliciting feedback on this topic: using git bisect to find "the commit that broke it." I am particularly interested in anyone who has automated the process so that it will build, test, pull, build, test, pull, etc. automatically without human intervention. If you've done anything like that please let us know. I'd like to see git bisect used more for people who are reporting bugs with phrases like, "It stopped working when I updated so I went back to the old git version and it works fine there." Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Yesterdays build: switch_rtp.c:3119 Activating RTP audio ICE is crashing clients
by Tim Meade 23 Apr '13

23 Apr '13

I updated serveral of our servers to yesterdays build. SOME of them cause the app to crash on calling out. I'm seeing switch_rtp.c:3119 Activating RTP audio ICE in the cli. The clients are complaining about bad ICE settings as well. Is this something in the latest builds? Thanks Tim

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sip to h323 video call via freeswitch
by Navnath Sonavne 23 Apr '13

23 Apr '13

Hi All, I am running freeswitch successfully on centos for audio and video call using xlite sip phones. Now I am trying different things. I am able to make audio call from X-Lite softphone(sip protocol endpoint) to Ekiga softphone(h323 protocol endpoint) via freeswitch. But my video call is not showing videos at both ends. So I wonder is this type of video call(sip softphone--->freeswitch--->h323 softphone) possible? If yes then please give me some advice to make it work. Regards, Navnath.

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registered end-point user agent info filter
by Henry Huang 23 Apr '13

23 Apr '13

H i, I have already tried to look things up in FreeSWITCH wiki or just googling, but didn't find it. Is there a way I can use condition in the dialplan to filter for destination extension User Agent name so I can set or disable proxy-media on the fly? I know I can get this information by doing the following in console, but don't know how to get the same information in dialplan and have it filtered. >sofia status profile interal reg Thanks, Henry

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[ERR] mod_sofia.c:3492 Invalid Gateway after reconnect
by k_petrovich＠mail.ru 23 Apr '13

23 Apr '13

Hello, after reconnect my internet. i'm get error, when i'm dialing outside. Strings from my log: 2013-04-23 09:48:46.478625 [DEBUG] switch_core_state_machine.c:157 sofia/internal/215@192.168.0.1:5060 Standard EXECUTE EXECUTE sofia/internal/215@192.168.0.1:5060 set(outside_call=true) 2013-04-23 09:48:46.478625 [DEBUG] mod_dptools.c:817 sofia/internal/215@192.168.0.1:5060 SET [outside_call]=[true] EXECUTE sofia/internal/215@192.168.0.1:5060 set(ringback=%(2000,4000,440.0,480.0)) 2013-04-23 09:48:46.478625 [DEBUG] mod_dptools.c:817 sofia/internal/215@192.168.0.1:5060 SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE sofia/internal/215@192.168.0.1:5060 bridge(sofia/gateway/golden50/3872424) 2013-04-23 09:48:46.478625 [ERR] mod_sofia.c:3492 Invalid Gateway 2013-04-23 09:48:46.478625 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW] 2013-04-23 09:48:46.478625 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY 2013-04-23 09:48:46.478625 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY 2013-04-23 09:48:46.478625 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY 2013-04-23 09:48:46.478625 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep 2013-04-23 09:48:46.478625 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] When i'm restart service, Freeswitch working ok.

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SIP regsitration timeouts
by Alex Lake 23 Apr '13

23 Apr '13

We're trying to understand SIP registration timeouts. When we do a sofia status profile internal reg, one of the lines includes something like this: Status Registered(UDP)(unknown) EXP(2013-04-22 17:22:00) EXPSECS(1796) It seems that when a phone registers, it says how long Freeswitch should retain the registration for (in this case there's half an hour left). What is the relationship between this mechanism and the one in config of the sip profile: <param name="session-timeout" value="1800"/> Or is this a totally different kind of session?

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