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Presence Not Working After Version 1.0.5pre9?
by Jerry Richards 24 Jan '10

24 Jan '10

Does anyone know why I do not see NOTIFY messages with presence status being sent out from FS for two Bria softphones? It used to work in my old version 1.0.5pre9. I upgraded Freeswitch on Jan 12 and Jan 20 and edited in my XML configurations, but I do not see the NOTIFY messages since version 1.0.5pre9. I have mostly default configuration and I added the manage-presence=true setting. I posted a pastebin at http://pastebin.freeswitch.org/11867. Can anyone tell why this isn't working? Best Regards, Jerry

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INCOMPATIBLE_DESTINATION?
by Sergey Okhapkin 24 Jan '10

24 Jan '10

I often see in FS log the following problem (bypass_media=true), SVN r16340: SDP sent out to gateway (INVITE): v=0 o=bandx-msw3 0 0 IN IP4 213.166.9.4 s=sip call c=IN IP4 213.166.9.6 t=0 0 m=audio 56032 RTP/AVP 0 8 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=silenceSupp:on - - - - SDP response from gateway (183 Session Progress): v=0 o=- 3473087019 3473087037 IN IP4 67.203.64.182 s=- c=IN IP4 67.203.64.182 t=0 0 m=audio 14116 RTP/AVP 0 a=sendrecv a=ptime:20 a=rtpmap:0 PCMU/8000 Freeswitch hangs up the call with hangup cause INCOMPATIBLE_DESTINATION. Where is incompatibility? There is common codec 0.

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Managing Presence on Gateways
by afshin afzali 24 Jan '10

24 Jan '10

Hi Guys, In the external profile (as in the internal) there is an option to enable presence functionality (with setting it to passive). My question is how does it mean presence functionality for a gateway which interfaces home domain to another one? Does it mean that the gateway could subscribe itself for some presence information in that domain in behaves of local users and relays them? Appreciate all comments, -- afshin

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Eavesdrop when using simring
by Wellie Chao 24 Jan '10

24 Jan '10

I have eavesdrop working fine on outbound calls and also inbound calls where there is a single DID per IP phone. When I have a DID that rings multiple extensions simultaneously, what is the best way to obtain information about which extension has picked up the call and store that using hash? I can set a variable before I issue the bridge action, like so: <action application="hash" data="insert/${domain_name}-spymap/646xxxyyyy-1000/${uuid}"/> <action application="bridge" data="user/1001@${domain_name},user/1002@${domain_name},user/1003@${domain_name},user/1004@${domain_name}"/> However, that doesn't tell me who actually picked up, so at best I can allow users to eavesdrop on the last incoming call to the main DID, not the last incoming call to a particular extension. Is there something I can do in the bridge that will cause it to set a variable once it knows which extension has picked up the call?

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Possibly Bug in mod_sofia
by afshin afzali 23 Jan '10

23 Jan '10

Hi, In sofia_reg.c line 190 : nua_handle_bind(gateway_ptr->nh, gateway_ptr->sofia_private); It seems the right handle be gateway_ptr->sub_nh. It causes every SUBSCRIBE message's response print "Gateway information missing" error. -- afshin

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Call for help on FSComm
by João Mesquita 23 Jan '10

23 Jan '10

FreeSWITCH™ers, I believe that everyone already heard about FSComm by now. Development is moving fast and features are being added very fast. Nonetheless, we need help to get this software the way we want. If you are a GUI Designer or have any design skills and want to contribute, please, get in touch with me by email, IM, IRC on #fscomm or even pigeon if you like. I believe everyone wants a sexy looking softphone with all the features FreeSWITCH™ is able to provide. Regards, João Mesquita FSComm Developer GTalk: jmesquita(a)gmail.com PayPal: jmesquita(a)gmail.com IRC: jmesquita on #fscomm @freenode

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CDR Gateways
by FERNANDO VILLARROEL 22 Jan '10

22 Jan '10

Dear All. I have defined various gateways in ~/sip-profiles/external My questions is if a gateway named foo that send calls from diferents IP address like x.x.x.x and x.x.x.y. How i can doing a group for both ip address where i can doing later a sql like accountcode=foo? In this moment if i need know the traffic of customer foo, i should doing two differents cdr, one for every ip address. It's fine or i can doing of different way? I hope anyone could me explain how i can doing. my gateway foo.xml <include> <gateway name="foo"> <param name="username" value="foo"/> <param name="password" value="foo"/> <param name="realm" value="x.x.x.x"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> </include> foo1.xml <include> <gateway name="foo1"> <param name="username" value="foo1"/> <param name="password" value="foo1"/> <param name="realm" value="x.x.x.y"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> </include> Both gateways foo and foo1 are the same customer my cdr_csv.conf.xml <configuration name="cdr_csv.conf" description="CDR CSV Format"> <settings>   <param name="default-template" value="sql"/>   <param name="rotate-on-hup" value="false"/>  <param name="legs" value="ab"/> </settings> <templates> <template name="sql">INSERT INTO v_cdr (v_id, caller_id_name, caller_id_number, destination_number, context, start_stamp, answer_stamp, end_stamp, duration, billsec, hangup_cause, uuid, bleg_uuid, accountcode, read_codec, write_codec, remote_media_ip, network_addr) VALUES ("{v_id}", "${caller_id_name}" ,"${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}" ,"${uuid}","${bleg_uuid}", "${accountcode}", "${read_codec}", "${write_codec}", "${remote_media_ip}", "${network_addr}");</template> The argument accountcode on my database is Blank or None for all records of gateways. Regards.

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 43, Issue 187
by Jerry Richards 22 Jan '10

22 Jan '10

I tried sending this Email earlier in the week but it got kicked back by our Mail Server, so if this is a duplicate, my apologies... Are you still working toward a release version 1.0.5? Best Regards, Jerry

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Choppy conference audio
by Tim St. Pierre 22 Jan '10

22 Jan '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello all, I'm having a tough time sorting out the audio in mod_conference, I'm hoping someone can point me in the right direction. I took a good look around the wiki, but couldn't really find what I was looking for. I have a test system set up, with three Aastra 9133i, a 6731i, and a 57i. All register, call fine, etc. Music on hold audio is perfect. Calls between phones are perfect. CPU load on the lab machine is about 8%, so not really busy. Conference audio is choppy, in the way that things get choppy when there is some sort of timing issue. Even with one member in the conference, the prompts and moh sound choppy. I played around with the rate and interval values, there wasn't any change. I tried disabling the monotonic timer, and that didn't change anything either. I'm running FreeBSD 7, with the latest release of FS downloaded last week. Thanks for any help. Let me know if I need to post any other information. - -Tim -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.4 (FreeBSD) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQCVAwUBS1oK2YqVcvNCnHOrAQLriAP/csjIfD/VP0CA3ePyRBKXbDPfxEiqx/cP iWILZ65F0bxaryKTYT2ZV8W7u+wH6YxMCaoej+H+yd1XG08hZr4kLUcewUaCTMna S8Zd24ReknEmL8d0AXCMgospf2wmwZZx2pJMFmbPN3IlXup20dKWOESBp/Dru3vL 63GE8s6AdyA= =ndAy -----END PGP SIGNATURE-----

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skypiax and xml-curl
by Peter P GMX 22 Jan '10

22 Jan '10

Hello, is there a way to manage skypiax via XML-curl besides the dialplan? Best regards Peter

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