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Re: [Freeswitch-users] sips dial string question
by Kristian Kielhofner 05 Aug '10

05 Aug '10

Heh... I love that I got a verbal response in person at almost exactly this same time yesterday. Cluecon rocks!!! -- Kristian Kielhofner http://blog.krisk.org ------------------------------ *From*: freeswitch-users-bounces(a)lists.freeswitch.org < freeswitch-users-bounces(a)lists.freeswitch.org> *To*: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> *Sent*: Thu Aug 05 09:40:58 2010 *Subject*: Re: [Freeswitch-users] sips dial string question It does but their might be cases that aren't handled properly. Since Aastra is the only phone I know of that does it like that and testing was limited. /b On Aug 4, 2010, at 10:24 AM, Kristian Kielhofner wrote: Does freeswitch support the sips uri scheme? -- Kristian Kielhofner http://blog.krisk.org

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Re: [Freeswitch-users] sips dial string question
by Kristian Kielhofner 05 Aug '10

05 Aug '10

Does freeswitch support the sips uri scheme? -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Sent: Wed Aug 04 10:07:31 2010 Subject: Re: [Freeswitch-users] sips dial string question do this sofia/blah@blah;transport=tls /b On Aug 4, 2010, at 2:19 AM, Seven Du wrote: > Hi, > > I'm trying tls between 2 freeswitches, one as a server and one as a > softphone. > > I think I successfully configured the server side, and copied the pem > files to my client side. > > > server: port 53 for sip and 54 for sips. > client: external profile port 5080 for sip and 5081 for sips. > > I can successfully call the server from client: > > bgapi originate sofia/external/sips:xxxx@my-server.com:54 &echo > > questions: > > on client side, how to setup > > 1) register a gateway to server with sips. I tried > realm=my-server.com:54 but doesn't seemed to work, is there any other > param I should set ? > I can register with sip(port 53). > > 2) then, can I use dial string like sofia/gateway/gw/xxxx with sips ? > > Thanks. > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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sips dial string question
by Seven Du 05 Aug '10

05 Aug '10

Hi, I'm trying tls between 2 freeswitches, one as a server and one as a softphone. I think I successfully configured the server side, and copied the pem files to my client side. server: port 53 for sip and 54 for sips. client: external profile port 5080 for sip and 5081 for sips. I can successfully call the server from client: bgapi originate sofia/external/sips:xxxx@my-server.com:54 &echo questions: on client side, how to setup 1) register a gateway to server with sips. I tried realm=my-server.com:54 but doesn't seemed to work, is there any other param I should set ? I can register with sip(port 53). 2) then, can I use dial string like sofia/gateway/gw/xxxx with sips ? Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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200 OK in tcpdump, but not anything in FS
by Mikhail Krivushin 04 Aug '10

04 Aug '10

Hello, everyone! We have problem - leg B answer with 200 OK, but not anything appear in log and appear in tcpdump. -- С уважением, Кривушин Михаил г. Томск сот. +7 927 728 6799 icq: 218 744 127 xmpp: KrivushinME(a)jabber.ru skype: mkrivushin

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Custom NOTIFY via a proxy
by Neil Burgess 04 Aug '10

04 Aug '10

Hi, 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 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), and use fs_path to push the call via the right load balancer. 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! Maybe another combination I am missing! Example packet capture follows: 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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Event queue is full
by Kozak Vladimir 04 Aug '10

04 Aug '10

Hi everybody. Can you tell me, please, what is the reason and how to avoid the error "Event queue is full!" FS log: 2010-08-03 14:16:48.213324 [DEBUG] sofia.c:4139 Channel sofia/external-subnet-10/sipp@10.122.246.14:5061 entering state [completed][200] 2010-08-03 14:16:48.213324 [DEBUG] switch_core_session.c:638 Send signal sofia/external-subnet-10/sipp@10.122.246.14:5061 [BREAK] 2010-08-03 14:16:48.213324 [WARNING] switch_event.c:1169 queued event at a lower priority 2/0! 2010-08-03 14:16:48.213324 [NOTICE] mod_dptools.c:717 Channel [sofia/external-subnet-10/sipp@10.122.246.14:5061] has been answered 2010-08-03 14:16:48.213324 [WARNING] switch_event.c:1169 queued event at a lower priority 2/0! EXECUTE sofia/external-subnet-10/sipp@10.122.246.14:5061 event(Event-App-Type=BP-STARTED,SYSTEM-ID=starpound,variable_sip_from_uri=sipp@10.122.246.14:5061,variable_sip_to_uri=6783257895@10.122.246.112:5260,variable_sip_req_user=6783257895) 2010-08-03 14:16:48.213324 [WARNING] switch_event.c:1169 queued event at a lower priority 2/0! 2010-08-03 14:16:48.213324 [WARNING] switch_event.c:1169 queued event at a lower priority 2/0! 2010-08-03 14:16:48.213324 [WARNING] switch_event.c:1169 queued event at a lower priority 2/0! EXECUTE sofia/external-subnet-10/sipp@10.122.246.14:5061 park() 2010-08-03 14:16:48.213324 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.213324 [DEBUG] sofia.c:4139 Channel sofia/external-subnet-10/sipp@10.122.246.14:5061 entering state [ready][200] 2010-08-03 14:16:48.313194 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.414213 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.514290 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.614241 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.714309 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.774320 [DEBUG] sofia.c:4139 Channel sofia/external-subnet-10/sipp@10.122.246.14:5061 entering state [terminated][408] 2010-08-03 14:16:48.774320 [NOTICE] sofia.c:4775 Hangup sofia/external-subnet-10/sipp@10.122.246.14:5061 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] 2010-08-03 14:16:48.774320 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.784288 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.814294 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.874289 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.884292 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.924295 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.987259 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.987259 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:49.024356 [CRIT] switch_event.c:1175 Event queue is full!

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Callback - A made call to B by web
by Shamun toha md 04 Aug '10

04 Aug '10

Dear Sir, How can i make CallBack feature? Scenario GUI: +----------------------------------------+ | FreeSwitch - Web/Mobile/Desktop portal | |========================================| | | | Call from: __+3216000000___________ | | | | Call To: __+4414000000___________ | | | | [ CALL ] [ Call at 8AM ] | | | +----------------------------------------+ * Call from == my mobile number * Call to == where i want to call Question. How can we do this using FreeSwitch? Also to have A-leg and B-leg billing to be 2 seperate Thanks Regards

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Re: [Freeswitch-users] answer machine
by Madovsky 03 Aug '10

03 Aug '10

found this thread http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg17343.… but is it relevant yet ? Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, August 03, 2010 6:50 PM Subject: answer machine Is there a way to set up the volume of answer machine ? Thanks F

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answer machine
by Madovsky 03 Aug '10

03 Aug '10

Is there a way to set up the volume of answer machine ? Thanks F

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getDigits() over ESL
by Chaitanya Bhatt // Viva 03 Aug '10

03 Aug '10

Hey I want to get DTMF digits over ESL. I got function "uuid_recv_dtmf" in freeswitch api document, but i am not getting how to retrieve data from this function. Can someone guide me about API command or function of get DTMF? Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya(a)vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009

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