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Custom SIP Notify via proxy
by Neil Burgess 23 Aug '10

23 Aug '10

Hello, Sent this post a couple of weeks back. Still trying to find a solution, so trying again and see if anyone has any thoughts! I am trying to send CUSTOM NOTIFY events to devices, using the ESL interface with a PHP script of the following form. I am having partial success, but coming up against problems when using with an outbound SIP proxy:- $e = new ESLevent("NOTIFY"); $e->addHeader("profile", "internal"); $e->addHeader("from", "1020"); $e->addHeader("to-uri", "sip:1019@192.168.2.3:5063;fs_path=sip:82.24.214.226:5063"); $e->addHeader("from-uri", "sip:1020@pbx.rm.com"); $e->addHeader("host", "pbx.rm.com"); $e->addHeader("event-string", "check-sync"); $e->addHeader("content-type", "application/simple-message-summary"); $e->addBody("ok"); $res = $sockFSServerCommand->sendEvent($e); In our setup, we have several FreeSwitch boxes sitting behind two OpenSIPS load balancers. These Load Balancers handle registrations, etc. When we originate calls from FreeSwitch, we use an originate URL of the form (sofia/internal/1001@sip.redmatter.com;fs_path=sip:84.45.30.2<mailto:sofia/internal/1001@sip.redmatter.com;fs_path=sip:84.45.30.2>), and use fs_path to push the sip call out via the right load balancer/ proxy. This works perfectly. So, with the custom NOTIFY's I also need to make them follow the same path. As mentioned, we already have the contact string in our hand from the OpenSIPS registration server, but can't see how I can use that plus force the outbound path for the Notify. Was thinking I could use fs_path in the uri (as above) but doesn't seem to work, i.e. the wrong IP is used, and NOTIFY is messed up somewhat (see example below)! Maybe another combination I am missing! ------------------------ NOTIFY sip:1019@192.168.2.3:5063;fs_path=sip:82.24.214.22:5063 SIP/2.0 Via: SIP/2.0/UDP 212.85.24.245;rport;branch=z9hG4bKv268QHcy539yc Max-Forwards: 70 From: <sip:1020@pbx.rm.com>;tag=0rer8BS2etmmc To: <sip:1019@192.168.2.3:5063>;fs_path=sip:82.24.214.22:5063 Call-ID: 5a08d5c6-1a8e-122e-3885-0024e86c401a CSeq: 134921 NOTIFY Contact: <sip:mod_sofia@212.85.24.245:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.5-20100211-0400-16602 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces Event: check-sync Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=timeout Content-Type: application/simple-message-summary Content-Length: 2 ok ------------------ Any thoughts gratefully appreciated. Rgds, Neil

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bypass media after fifo bridge
by Rong Shen 23 Aug '10

23 Aug '10

Hi, We've been using mod_fifo to connect calls for a while, it has all worked well except we are never able to bypass the media after a consumer and caller are connected. Enabling bypass_media_after_bridge in caller's dailplan doesn't seem to pass the variable to the point where the caller is popped out of the queue. We also tried to set bypass_media_after_bridge in the consumer's session, and use variable_fifo_consumer_caller_import to import it into the caller's session, but it didn't seem to have any effect - media is still going through the freeswitch (an event dump attached). Any idea how to set the bypass media option when using mod_fifo? Thanks. Regards, Rong

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mod_skinny issue
by Terry Moore-Read 22 Aug '10

22 Aug '10

When starting freeswitch, I'm seeing the following : 2010-08-05 23:08:43.665581 [ERR] mod_skinny.c:357 SQL ERR: [DELETE FROM skinny_devices] no such table: skinny_devices 2010-08-05 23:08:43.666177 [ERR] mod_skinny.c:357 SQL ERR: [DELETE FROM skinny_lines] no such table: skinny_lines 2010-08-05 23:08:43.666761 [ERR] mod_skinny.c:357 SQL ERR: [DELETE FROM skinny_buttons] no such table: skinny_buttons 2010-08-05 23:08:43.667323 [ERR] mod_skinny.c:357 SQL ERR: [DELETE FROM skinny_active_lines] no such table: skinny_active_lines It appears from mod_skinny.c that these tables should have been created by this point if they don't exist. I'm not sure why this isn't happening. FreeSWITCH Version 1.0.head (git-b60d6b3 2010-08-05 16-07-14 -0400) using sqlite

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Does FS ESL provide session command?
by jesse 22 Aug '10

22 Aug '10

According to http://wiki.freeswitch.org/wiki/Event_Socket, ESL only supports: api, bapi, event, myevents, divert_events, filter,sendevent, sendmsg, hangup, auth, exit,log, nolog, noevents, nixevent. How come I don't see a way to create a session via ESL. like in lua, freeswitch.Session(dialstring); one way to accomplish this in ESL is : api originate dialstring &hold(), is this = Session(dialstring) ? thanks

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Mod_h323 fail to load
by Sadjad Seyed-Ahmadian 22 Aug '10

22 Aug '10

Dear All, I try to install mod_h323, I compile it as it comes in wiki (http://wiki.freeswitch.org/wiki/Mod_h323) but when I try to load mod_h323 in fs_cli I faces this error: > >2010-08-20 20:06:07.781508 [CRIT] switch_loadable_module.c:882 Error Loading >module /usr/local/freeswitch/mod/mod_h323.so >**libh323_linux_x86_.so.1.22.0: cannot open shared object file: No such file or >directory** > >I cannot find what the problem is. and also I cannot find libh323_linux file. >Would somebody please help me with it?

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how to answer skype call after recent change?
by xuyan yang 22 Aug '10

22 Aug '10

Hi, I have met a problem that skype to skype call become silent when eavesdrop is working. And already reported to JIRA. When trying to test with latest git head, SKYPE never answer any call. It seems to be a recent upgrade which has not been documented. How should I change the config to let skype answer all calls as before?

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Half calls end up in echo mode without counter-party's sound, since 26Jul git version
by xuyan yang 21 Aug '10

21 Aug '10

I have a IVR written in js running on git version. During the test in Jul it works fine. But recently, I find that nearly half calls end up in silence and only echoing its own sound. Checked the recording on server, it is the same. IVR's speech is played back to the FS instead of the caller. The version is compiled on 26jul git. The same problem exist for both of my SIP providers and skypeopen, So it should not be the providers' fault. The git tree today is also tested with the same problem. But when I roll back to 1.0.6 everything back to normal. As I am updating the version quite often in Jul, this is very likely to be caused by some changes just before 26Jul. Does anyone using git version after 26Jul have the same problem?

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[FreeSWITCH] How to register SIP users with FreeSWITCH using a client.
by Tharindu Madushanka 21 Aug '10

21 Aug '10

Hi, I have built a SIP client for my University project and I could make SIP calls between two accounts using third party free SIP accounts. I used sip2sip.org to test the functionality. Now I would like to install a SIP server at my local machine and test how it works. Then I was looking for SIP open source projects and found this project. What I would like to do is. from my client app I want to register ( sign up ) for SIP accounts with FreeSWITCH server I installed.. Could somebody help me to give some points what I have to look into achieve this functionality.. It's really useful.. Thank you and Kind Regards, Tharindu Madushanka tharindufit.wordpress.com

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Webapi Examples location
by Phone 21 Aug '10

21 Aug '10

I am exploring the Webapi. The sample applications "Voicemail" and "Telecast" are working as they should, but where do I find the source for them?? Thanks

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simple echo failed
by jesse 21 Aug '10

21 Aug '10

What the heck is wrong with my system even simple "originate user/1001 & echo" doesn't work???? freeswitch@xyz-host> originate user/1001 & echo 2010-08-21 00:00:29.647778 [NOTICE] switch_channel.c:776 New Channel sofia/internal/sip:1001@192.168.2.111:16176 [21277012-acc5-464c-b415-595636e5ad73] 2010-08-21 00:00:29.892892 [INFO] sofia.c:662 sofia/internal/sip:1001@192.168.2.111:16176 Update Callee ID to "1001" <1001> 2010-08-21 00:00:29.895316 [NOTICE] sofia.c:4389 Ring-Ready sofia/internal/sip:1001@192.168.2.111:16176! 2010-08-21 00:00:31.727349 [WARNING] sofia_reg.c:387 google.com Failed Registration, setting retry to 30 seconds. 2010-08-21 00:00:33.830453 [NOTICE] sofia.c:4875 Channel [sofia/internal/sip:1001@192.168.2.111:16176] has been answered 2010-08-21 00:00:33.830453 [NOTICE] switch_ivr.c:1475 Transfer sofia/internal/sip:1001@192.168.2.111:16176 to echo[&@default] +OK 21277012-acc5-464c-b415-595636e5ad73 2010-08-21 00:00:33.831470 [NOTICE] switch_core_state_machine.c:131 Hangup sofia/internal/sip:1001@192.168.2.111:16176 [CS_ROUTING] [NO_ROUTE_DESTINATION] freeswitch@xyz-host> 2010-08-21 00:00:33.837480 [NOTICE] switch_core_session.c:1193 Session 31 (sofia/internal/sip:1001@192.168.2.111:16176) Ended 2010-08-21 00:00:33.837480 [NOTICE] switch_core_session.c:1195 Close Channel sofia/internal/sip:1001@192.168.2.111:16176 [CS_DESTROY] thanks! jesse

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