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SIP 100 response and media
by Boris Kovalenko 07 Jul '11

07 Jul '11

Hello! This thread again. There are two logs. First between Cisco<->Cisco call and second between Cisco<->FreeSwitch. Cisco<->Cisco works fine. As I can see the difference is that Cisco has a=rtpmap:8 PCMA/8000 and FreeSwitch not. May be this is a problem? Is this field mandatory? Jul 6 18:20:39.341: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:28032@192.168.35.15:5060 SIP/2.0 Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F Remote-Party-ID: "test" <sip:30030@10.240.14.15>;party=calling;screen=no;privacy=off From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA(a)10.240.14.15 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2076058060-2805141984-2507440228-2131084746 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1309976439 Contact: <sip:30030@10.240.14.15:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 187 v=0 o=CiscoSystemsSIP-GW-UserAgent 0 703 IN IP4 10.240.14.15 s=SIP Call c=IN IP4 10.240.14.15 t=0 0 m=audio 18720 RTP/AVP 8 c=IN IP4 10.240.14.15 a=rtpmap:8 PCMA/8000 a=ptime:20 Jul 6 18:20:39.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA(a)10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Jul 6 18:20:39.369: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15>;tag=CBEFC0-16E4 Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA(a)10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: <sip:28032@192.168.35.15:5060> Supported: replaces Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 194 When call is between Cisco<->Freeswitch messages are Jul 6 18:21:11.966: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:73435230030@192.168.3.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp Max-Forwards: 67 From: "anonymous" <sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To: <sip:73435230030@192.168.3.6:5060> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Contact: <sip:mod_sofia@192.168.1.1:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1c95ad9 2011-01-20 22-43-50 -0300 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 329 X-FS-Support: update_display Remote-Party-ID: "anonymous" <sip:0000000000@192.168.1.1>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1309956111 1309956112 IN IP4 192.168.1.1 s=FreeSWITCH c=IN IP4 192.168.1.1 t=0 0 m=audio 20360 RTP/AVP 8 0 18 101 13 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 19030 RTP/AVP 31 34 98 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 H264/90000 Jul 6 18:21:11.986: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp From: "anonymous" <sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To: <sip:73435230030@192.168.3.6:5060> Date: Wed, 06 Jul 2011 18:21:11 GMT Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 -- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005

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Custom GUI for FS
by Alessandro 07 Jul '11

07 Jul '11

Hi, I'm going to develop a custom GUI web-based for freeswitch that allow only to add/remove extension and show the call active and the status of extensions configured. Where can I find in FS the information about actives call and extensions status? Thanks Alessandro Luppi -- Ing. Alessandro Luppi Software development Seletech srl Via Collodi 8, 20052 Monza (MI) - Italy Tel: +39.039.5962000 - Fax: +39.039.9716905 email: a.luppi(a)seletech.com - Web: www.seletech.com or www.seletech.eu

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Chinese sounds
by Seven Du 07 Jul '11

07 Jul '11

Hi, We got someone in the Chinese Community would like to record a Chinese version of sounds, here's some questions for the community - 1) Where can I find the original English scripts ? It would be hard to listen all sound files 2) What's base256/*.wav usage? 3) We'd like to share to world, so, how to upload to files.freeswitch.org when we are done? 4) I guess the raw recording should be 48K, 16bit, single channel, any other params should we set when recording? Thanks. -- Seven Du About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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question about continue_on_fail
by Boris Kovalenko 07 Jul '11

07 Jul '11

Hello! Here is an extension: <extension name="ext_domestic_e164"> <condition field="destination_number" expression="^(7[0-9]{10})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/>  <action application="lcr" data="$1 lcr_tagnet"/> <action application="set" data="call_timeout=60"/> <action application="set" data="bridge_answer_timeout=60"/> <action application="set" data="disable-transcoding=true"/> <action application="bridge" data="${lcr_auto_route}"/> </condition> </extension> lcr_auto_route returns 2 bridges to a same destination. [lcr_carrier=naukanet.ru,lcr_rate=1.45140,v_ats_dstport=50002]sofia/epbx/89979049898ZZZ@X.X.X.42:5061|[lcr_carrier=mtt.ru,lcr_rate=1.60480,v_ats_dstport=50006]sofia/epbx/79049898ZZZ@Y.Y.Y.132:5060 Here is a problem: when the reject from the remote is [CS_CONSUME_MEDIA] [USER_BUSY] the call is routed to the next bridge. When the reject is [CS_EXCHANGE_MEDIA] [USER_BUSY] the bridge hangs up. So, what is the right way to stop processing when USER_BUSY arrived? I tried continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION without success. -- Regards, Boris

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mod dingaling, gv incoming calls dialplan
by Avi 07 Jul '11

07 Jul '11

Hello all. I apologize for the newbie question but after much reading, I can't seem to figure out what the incoming dialplan for GV/extension needs to look like. Currently I have what is listed in the twiki page for dingaling and google voice. However there is a line in the twiki page that says "Put the following in your dialplan (same context as the receiving extension)". I'm not sure what the receiving extension should look like since I can't seem to find an example of that. Here is what I'm seeing in the logs http://pastebin.com/7j5e7qTm Since I have nothing for the extension 2001, I presume the call hangs up on the FS end. The extension never rings. Any help is appreciated and I sure hope I get some. FYI, outgoing calls seems to be working just fine. Thanks much.

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Recording - one leg of the call goes faster than the other
by Kees Varekamp 07 Jul '11

07 Jul '11

Hi all, We are having a weird issue with call recordings: one leg of the call goes around 1 second per minute faster than the other. At the end of a 10 minute recording this means that answers come around 10 seconds ahead of the corresponding questions. This is on FreeSWITCH Version 1.0.head (git-38f06a3 2011-05-19 15-39-43 -0500) This happens pretty much on all the recordings we do. Found this similar sounding problem on the bug tracker: http://jira.freeswitch.org/browse/FS-2532. Has anyone experienced this same problem? Is there a solution? Thanks, Kees

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Ok I gotta share this...
by Ken Rice 06 Jul '11

06 Jul '11

Looks like Lanman and some other guys at Cuda had an Office Space minute http://www.youtube.com/watch?v=CGL43sk16XE K

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Call Dropped When Answered If Bridged To Internal Extension And PSTN Number Via
by Jerry Richards 06 Jul '11

06 Jul '11

Hello, If I bridge a call to both an internal extension and a PSTN number (via FreeTDM), and then answer the call at the internal extension, the call is dropped. I posted the log at http://pastebin.freeswitch.org/16634 The log shows many errors of the type: reading on a session with no media! I have bypass-media 'true' (not sure if this has anything to do with it). Any clue why this is happening? Thanks, Jerry

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ClueCon News: Free OpenSIPS Training, Sponsor Logo - Last Chance, Party RSVP
by marketing＠cluecon.com 06 Jul '11

06 Jul '11

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FreeSWITCH Conference Call Today
by Michael Collins 06 Jul '11

06 Jul '11

C'mon down! http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_06 Lots of new stuff to talk about! -MC

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