October 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Expression in dialplan
by Ahmed Munir 22 Oct '09

22 Oct '09

Thanks for reply. Actually I want to match values and strings. Like IP_CALL=="true" or IP_CALL != "true" or Date >= "22/10/09" and vice versa. Last thing I forgot to mention, how can I implement condition in dial plan xml when using operation and/or i.e. IP_CALL == "false" and PST_CALL == "false" / IP_CALL == "false" or PST_CALL == "false". Note: I only want to implement it in dial plan not in any script. Kindly advise the solution. ---------- Forwarded message ---------- From: Michael Collins <msc(a)freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 21 Oct 2009 21:36:31 -0700 Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179 On Wed, Oct 21, 2009 at 9:28 PM, Mark Campbell-Smith < mcampbellsmith(a)gmail.com> wrote: > Can't you use the inline statement to set a variable so that it can be > used directly in a condition? > > http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions > > Yes you can. Just don't abuse it like Tony said. If you find yourself doing it a lot then it's a sure sign that you're taking the wrong approach. -MC ---------- Forwarded message ---------- From: Michael Collins <msc(a)freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 21 Oct 2009 21:41:31 -0700 Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179 On Wed, Oct 21, 2009 at 9:08 PM, Ahmed Munir <ahmedmunir007(a)gmail.com>wrote: > Hi, > > Thanks for reply, it really helped me. One more thing to ask, how can we > make decision against >,<, >=, <= in condition header? Like we use == for > action and != for anti-action. > > Kindly highlight it. > > You can only do greater than and less than in the date/time matching. See the date/time example in the default.xml dialplan file. You can also use regular expressions if you're in a pinch. For example, if you need to match numbers >= 1100 and <= 1500 you could just use this regex: ^(1[1234]\d\d|1500)$ The real question, though, is this: what types of values do you need to match for GT or LT? Date/time? Money? Other? That will determine if you need to use a script or just the dialplan. -MC

1 0

Call custom variable in condition
by Ahmed Munir 22 Oct '09

22 Oct '09

Hi, I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I want to use it in condition, as I'm listing down the configuration below; <context name="SIP_incoming"> <extension name="call-sip-extensions"> <condition field="destination_number" expression="^(\d+)$"> <action application="set" data="AUTHENTICATION_STATUS=0"/> <action application="transfer" data="${AUTHENTICATION_STATUS} XML Authen_Status"/> </condition> </extension> </context> <context name="Authen_Status"> <extension name="exten-auth-status"> <condition field="AUTHENTICATION_STATUS" expression="^0$"> <action application="answer"/> <action application="playback" data="play.wav"/> </condition> </extension> </context> But unfortunately it is not working. Kindly advise me how to do implement it(Note: I don't want to call script). And one more thing to ask how can I transfer the values within the same context? -- Regards, Ahmed Munir

6 6

Sofia SIP external profile gateway names
by Muhammad Shahzad 22 Oct '09

22 Oct '09

Hi, I am facing a problem with Sofia SIP external profile. Basically i have 10+ accounts, say a101 - a110, to a single service provider, say xyz.com. For each account a have created a gateway in external profile's directory i.e. /usr/local/freeswitch/conf/sip_profiles/external/a101.xml up to a110.xml. At first the problem was none of the accounts were registering since, FS was trying to send registration requests to host "a101" instead of xyz.com, but later when i set xyz.com as proxy and register proxy address, it successfully registered all accounts. Now i am facing a similar problem in dialplan, for example if i try to dialout via gateway a101, call immediately fails with NORMAL_TEMPORARY_FAILURE. When i trun on sip tracing i don't see any INVITE message sent to provider xyz.com. 2009-10-22 06:51:55.325393 [NOTICE] switch_channel.c:613 New Channel sofia/external/00923344224088@a101 [b7663caf-cd29-4213-9059-ae880d49b0ca] 2009-10-22 06:51:55.516382 [NOTICE] sofia.c:4039 Hangup sofia/external/00923344224088@a101 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] API CALL [originate(sofia/external/00923344224088@a101 9999)] output: -ERR NORMAL_TEMPORARY_FAILURE 2009-10-22 06:51:55.527381 [NOTICE] switch_core_session.c:1132 Session 2 (sofia/external/00923344224088@a101) Ended 2009-10-22 06:51:55.527381 [NOTICE] switch_core_session.c:1134 Close Channel sofia/external/00923344224088@a101 [CS_DESTROY] I think its trying to look up a101 instead of xyz.com to send INVITE. Kindly help. Thank you. -- ________________________________________________________ | | | FATAL ERROR --- O X | |_______________________________________________________| | You have moved the mouse. | | Windows must be restarted for the changes to take effect. | | <OK> | ####################################/ Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

1 0

CS_REPORTING Channel event state
by Dome Charoenyost 22 Oct '09

22 Oct '09

Dear All What's CS_REPORTING state ? I found many channels not hang up ans state is CS_REPORTING e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20 21:50:26,1256050226,sofia/external/xxxxxxxxxxxx@xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,xxxxxxxxxxxxxxxxxxxx,xx.xxx.xxx.xxx,7050,,,XML,public,,,,, BG Dome C.

3 2

NOT in dialplan expression
by Mark Campbell-Smith 22 Oct '09

22 Oct '09

Hi! How do I do a NOT equal to in a dialplan expression Normaly in regex I would use the ! character. This doesn't seem to work in FS.. ie <condition field="${variable}" expression="!^1"> Shouldn't that match when the variable is not starting with one?

6 7

Calls dropping on SIP timer expiry due to UPDATE's being ignored.
by Keith Laaks 21 Oct '09

21 Oct '09

Hi, Hope someone knows how I am able to get around this one. Here goes... Did an upgrade to trunk (from a July vintage build) last week and noticed calls out to a provider were now failing after about 30 seconds or so - post answer. Tried latest (15183) - same thing. Analysing, I see that I have multiple UPDATE messages now being sent to the provider, but no response being sent back to FS. So FS times out and eventually kills the call. Interestingly, it only drops the A-leg; the B-leg remains up till the B party hangs up. I cant recall seeing these UPDATE messages before... The intent of the UPDATE seems to be to send the callee name & number to the B-leg. If its the provider's sip stack that's broken w.r.t. handling UPDATE - is there any way to get around it by doing something in my config to ensure these UPDATE's are not 'triggered' ? Some traces below. Any suggestions welcomed... Best Regards Keith Pretoria, South Africa. -------------------------------------------------------------------------------------------------------------------------------------------------- send 1048 bytes to udp/[196.10.11.12]:5060 at 13:24:04.249269: ------------------------------------------------------------------------ INVITE sip:27835551111@196.10.11.12 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m Max-Forwards: 67 From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12> Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: <sip:gw+vprov@10.17.10.10:5060;transport=udp;gw=vprov> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-Actually-Support: UPDATE Remote-Party-ID: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10 s=FreeSWITCH c=IN IP4 10.17.10.10 t=0 0 m=audio 12862 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2009-10-21 15:24:04.248448 [DEBUG] sofia.c:3493 Channel sofia/vvrf/27835551111 entering state [calling][0] recv 601 bytes from udp/[196.10.11.12]:5060 at 13:24:04.307690: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: "vprov C5CM" <sip:196.10.11.12:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow-Events: talk Allow-Events: refer Content-Disposition: session X-Actually-Support: UPDATE Remote-Party-ID: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;party=calling;screen=yes;privacy=off Content-Length: 0 ------------------------------------------------------------------------ recv 879 bytes from udp/[196.10.11.12]:5060 at 13:24:08.508162: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: "vprov C5CM" <sip:196.10.11.12:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow-Events: talk Allow-Events: refer Content-Disposition: session X-Actually-Support: UPDATE Remote-Party-ID: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;party=calling;screen=yes;privacy=off Content-Type: application/sdp Content-Length: 233 v=0 o=Clarent 152602 152603 IN IP4 196.10.11.15 s=Clarent C5CM c=IN IP4 196.10.11.15 t=0 0 m=audio 5230 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ 2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3493 Channel sofia/vvrf/27835551111 entering state [proceeding][183] 2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3500 Remote SDP: v=0 o=Clarent 152602 152603 IN IP4 196.10.11.15 s=Clarent C5CM c=IN IP4 196.10.11.15 t=0 0 m=audio 5230 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2009-10-21 15:24:08.507506 [DEBUG] sofia_glue.c:3144 Audio Codec Compare [G729:18:8000:20]/[G729:18:8000:20] 2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:2102 Set Codec sofia/vvrf/27835551111 G729/8000 20 ms 160 samples 2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:3104 Set 2833 dtmf payload to 101 2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:2336 AUDIO RTP [sofia/vvrf/27835551111] 10.17.10.10 port 12862 -> 196.10.11.15 port 5230 codec: 18 ms: 20 2009-10-21 15:24:08.508561 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 160 bytes per 20ms 2009-10-21 15:24:08.509616 [NOTICE] sofia_glue.c:2771 Pre-Answer sofia/vvrf/27835551111! 2009-10-21 15:24:08.509616 [DEBUG] switch_channel.c:1992 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:08.510671 [DEBUG] switch_ivr_originate.c:615 Setting codec string on sofia/inetticky/27878050000(a)196.222.3.4 to G729@8000h@20i 2009-10-21 15:24:08.510671 [DEBUG] switch_ivr_originate.c:2326 sofia/inetticky/27878050000(a)196.222.3.4 receive message [PROGRESS] 2009-10-21 15:24:08.510671 [INFO] switch_ivr_originate.c:2326 Sending early media 2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:3144 Audio Codec Compare [G729:18:8000:0]/[G729:18:8000:20] 2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:2102 Set Codec sofia/inetticky/27878050000(a)196.222.3.4 G729/8000 20 ms 160 samples 2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:3104 Set 2833 dtmf payload to 101 2009-10-21 15:24:08.510671 [DEBUG] sofia_glue.c:2336 AUDIO RTP [sofia/inetticky/27878050000(a)196.222.3.4] 196.222.3.4 port 16840 -> 41.242.6.167 port 62936 codec: 18 ms: 20 2009-10-21 15:24:08.510671 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 160 bytes per 20ms 2009-10-21 15:24:08.511769 [NOTICE] sofia_glue.c:2771 Pre-Answer sofia/inetticky/27878050000(a)196.222.3.4! 2009-10-21 15:24:08.511769 [INFO] mod_sofia.c:1582 Ring SDP: v=0 o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4 s=FreeSWITCH c=IN IP4 196.222.3.4 t=0 0 m=audio 16840 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-10-21 15:24:08.511769 [DEBUG] switch_core_session.c:712 Send signal sofia/vvrf/27835551111 [BREAK] 2009-10-21 15:24:08.511769 [DEBUG] switch_core_session.c:653 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:08.511769 [DEBUG] sofia.c:407 sofia/vvrf/27835551111 receive message [DISPLAY] 2009-10-21 15:24:08.511769 [DEBUG] switch_ivr_originate.c:2368 Originate Resulted in Success: [sofia/vvrf/27835551111] 2009-10-21 15:24:08.511769 [DEBUG] switch_channel.c:182 sofia/vvrf/27835551111 receive message [AUDIO_SYNC] 2009-10-21 15:24:08.511769 [DEBUG] switch_channel.c:182 sofia/inetticky/27878050000(a)196.222.3.4 receive message [AUDIO_SYNC] 2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:975 sofia/vvrf/27835551111 receive message [BRIDGE] 2009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:653 Send signal sofia/vvrf/27835551111 [BREAK] 2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:982 sofia/inetticky/27878050000(a)196.222.3.4 receive message [BRIDGE] send 973 bytes to udp/[41.242.6.167]:62928 at 13:24:08.512974: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 41.242.6.167:56675;branch=z9hG4bK2234c0462cdad03fc318bd6c7438282;rport=629282009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:653 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:08.512832 [DEBUG] switch_ivr_bridge.c:1026 (sofia/vvrf/27835551111) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2009-10-21 15:24:08.512832 [DEBUG] switch_core_session.c:985 Send signal sofia/vvrf/27835551111 [BREAK] 2009-10-21 15:24:08.512832 [DEBUG] switch_core_state_machine.c:306 (sofia/vvrf/27835551111) Running State Change CS_EXCHANGE_MEDIA 2009-10-21 15:24:08.512832 [DEBUG] switch_core_state_machine.c:343 (sofia/vvrf/27835551111) State EXCHANGE_MEDIA 2009-10-21 15:24:08.512832 [DEBUG] mod_sofia.c:436 SOFIA LOOPBACK From: "Keith PhoneADSL" <sip:27878050000@196.222.3.4>;tag=3283506127 To: <sip:0835551111@196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc Call-ID: 4252084844@192_168_1_65 CSeq: 3 INVITE Contact: <sip:0835551111@196.222.3.4:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-Actually-Support: UPDATE v=0 o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4 s=FreeSWITCH c=IN IP4 196.222.3.4 t=0 0 m=audio 16840 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2009-10-21 15:24:08.512832 [DEBUG] sofia.c:3493 Channel sofia/inetticky/27878050000(a)196.222.3.4 entering state [early][183] recv 928 bytes from udp/[196.10.11.12]:5060 at 13:24:14.188781: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: "vprov C5CM" <sip:196.10.11.12:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow-Events: talk Allow-Events: refer Content-Disposition: session X-Actually-Support: UPDATE Remote-Party-ID: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;party=calling;screen=yes;privacy=off Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE Content-Type: application/sdp Content-Length: 233 v=0 o=Clarent 153288 153288 IN IP4 196.10.11.15 s=Clarent C5CM c=IN IP4 196.10.11.15 t=0 0 m=audio 5230 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 396 bytes to udp/[196.10.11.12]:5060 at 13:24:14.189032: ------------------------------------------------------------------------ ACK sip:196.10.11.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKZ07eFr88eH7Qg Max-Forwards: 70 From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 ACK Contact: <sip:gw+vprov@10.17.10.10:5060;transport=udp;gw=vprov> Content-Length: 0 ------------------------------------------------------------------------ 2009-10-21 15:24:14.188910 [DEBUG] sofia.c:3493 Channel sofia/vvrf/27835551111 entering state [ready][200] 2009-10-21 15:24:14.188910 [DEBUG] switch_channel.c:2105 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:14.188910 [NOTICE] sofia.c:3928 Channel [sofia/vvrf/27835551111] has been answered 2009-10-21 15:24:14.188910 [DEBUG] switch_channel.c:182 sofia/vvrf/27835551111 receive message [AUDIO_SYNC] 2009-10-21 15:24:14.204823 [DEBUG] switch_ivr_bridge.c:378 sofia/inetticky/27878050000(a)196.222.3.4 receive message [ANSWER] 2009-10-21 15:24:14.204823 [DEBUG] mod_sofia.c:543 Local SDP sofia/inetticky/27878050000(a)196.222.3.4: v=0 o=FreeSWITCH 1256114608 1256114610 IN IP4 196.222.3.4 s=FreeSWITCH c=IN IP4 196.222.3.4 t=0 0 m=audio 16840 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-10-21 15:24:14.204823 [DEBUG] switch_core_session.c:712 Send signal sofia/vvrf/27835551111 [BREAK] 2009-10-21 15:24:14.204823 [DEBUG] switch_core_session.c:653 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:14.204823 [NOTICE] switch_ivr_bridge.c:378 Channel [sofia/inetticky/27878050000(a)196.222.3.4] has been answered 2009-10-21 15:24:14.204823 [DEBUG] switch_channel.c:182 sofia/inetticky/27878050000(a)196.222.3.4 receive message [AUDIO_SYNC] send 983 bytes to udp/[41.242.6.167]:62928 at 13:24:14.205173: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 41.242.6.167:56675;branch=z9hG4bK2234c0462cdad03fc318bd6c7438282;rport=62928 From: "Keith PhoneADSL" <sip:27878050000@196.222.3.4>;tag=3283506127 To: <sip:0835551111@196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc Call-ID: 4252084844@192_168_1_65 CSeq: 3 INVITE Contact: <sip:0835551111@196.222.3.4:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-Actually-Support: UPDATE v=0 o=FreeSWITCH 1256114608 1256114609 IN IP4 196.222.3.4 s=FreeSWITCH c=IN IP4 196.222.3.4 t=0 0 m=audio 16840 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2009-10-21 15:24:14.204823 [DEBUG] sofia.c:3493 Channel sofia/inetticky/27878050000(a)196.222.3.4 entering state [completed][200] 2009-10-21 15:24:14.224959 [DEBUG] sofia.c:407 sofia/vvrf/27835551111 receive message [DISPLAY] send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:14.225162: ------------------------------------------------------------------------ UPDATE sip:196.10.11.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac Max-Forwards: 70 From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947387 UPDATE Contact: <sip:gw+vprov@10.17.10.10:5060;transport=udp;gw=vprov> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 247 P-Asserted-Identity: "unknown" <27835551111> v=0 o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10 s=FreeSWITCH c=IN IP4 10.17.10.10 t=0 0 m=audio 12862 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2009-10-21 15:24:14.224959 [DEBUG] sofia.c:3493 Channel sofia/vvrf/27835551111 entering state [calling][0] recv 733 bytes from udp/[41.242.6.167]:62928 at 13:24:14.533488: ------------------------------------------------------------------------ ACK sip:0835551111@196.222.3.4:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 41.242.6.167:56675;branch=z9hG4bKd2c91e1afbf0880332fbeba5fc9c04a6;rport From: "Keith PhoneADSL" <sip:27878050000@196.222.3.4>;tag=3283506127 To: <sip:0835551111@196.222.3.4>;user=phone;tag=g2DFFpS0QN4cc Call-ID: 4252084844@192_168_1_65 CSeq: 3 ACK Contact: <sip:27878050000@41.242.6.167:56675> Proxy-Authorization: Digest username="27878050000", realm="196.222.3.4", qop=auth, algorithm=MD5, uri="sip:0835551111@196.222.3.4", nonce="0092f192-be45-11de-a2c1-a70e5037c20e", nc=00000001, cnonce="ab2d49987adb74997fc6b236fc16d23", response="445ae74c1f910e45d9aa1043402b4dbb" Max-Forwards: 70 User-Agent: C455 IP021910000000 Content-Length: 0 ------------------------------------------------------------------------ 2009-10-21 15:24:14.533538 [DEBUG] sofia.c:3493 Channel sofia/inetticky/27878050000(a)196.222.3.4 entering state [ready][200] 2009-10-21 15:24:14.545161 [DEBUG] switch_core_session.c:712 Send signal sofia/vvrf/27835551111 [BREAK] 2009-10-21 15:24:14.545161 [DEBUG] switch_core_session.c:712 Send signal sofia/inetticky/27878050000(a)196.222.3.4 [BREAK] 2009-10-21 15:24:14.565231 [DEBUG] switch_ivr_bridge.c:122 sofia/vvrf/27835551111 receive message [DISPLAY] 2009-10-21 15:24:14.565231 [DEBUG] switch_ivr_bridge.c:122 sofia/inetticky/27878050000(a)196.222.3.4 receive message [DISPLAY] send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:14.727678: ------------------------------------------------------------------------ UPDATE sip:196.10.11.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac Max-Forwards: 70 From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947387 UPDATE Contact: <sip:gw+vprov@10.17.10.10:5060;transport=udp;gw=vprov> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 247 P-Asserted-Identity: "unknown" <27835551111> v=0 o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10 s=FreeSWITCH c=IN IP4 10.17.10.10 t=0 0 m=audio 12862 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ send 947 bytes to udp/[196.10.11.12]:5060 at 13:24:15.728255: ------------------------------------------------------------------------ UPDATE sip:196.10.11.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bK0907gKScctXac Max-Forwards: 70 From: "Keith PhoneADSL" <sip:878050000@10.17.10.10>;tag=Upa3NvXpBB1eF To: <sip:27835551111@196.10.11.12>;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947387 UPDATE Contact: <sip:gw+vprov@10.17.10.10:5060;transport=udp;gw=vprov> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 247 P-Asserted-Identity: "unknown" <27835551111> v=0 o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10 s=FreeSWITCH c=IN IP4 10.17.10.10 t=0 0 m=audio 12862 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 [Many more copies of the UPDATE's continue ... then ] 2009-10-21 15:24:46.228024 [DEBUG] sofia.c:3493 Channel sofia/vvrf/27835551111 entering state [terminated][408] 2009-10-21 15:24:46.228024 [NOTICE] sofia.c:4039 Hangup sofia/vvrf/27835551111 [CS_EXCHANGE_MEDIA] [RECOVERY_ON_TIMER_EXPIRE] 2009-10-21 15:24:46.228024 [DEBUG] switch_channel.c:1896 Send signal sofia/vvrf/27835551111 [KILL]

3 2

can't dial from IPv6 to IPv6
by ineya ineya 21 Oct '09

21 Oct '09

Hi, I installed the phonenix release, and changed IPv4, IPv6 addresses and SIP port. Directory and dialplan should be default ones. My phones are able to register when using just IPv4, and I can call from one to another. When I try the same thing with IPv6, the call to other phone fails with [ USER_NOT_REGISTERED], however I'm able to call 9999 and hear MOH. Here you can find the log: http://pastebin.freeswitch.org/10757 Freeswitch is running on 10.80.62.40/2000:2::1 phones are 2000:2::1001 and 2000:2::1002 Am I doing something wrong? Maybe I forgot to configure something, but I can't see what it is. Any hints are appreciated.

5 19

3rd Party Dial Plan Tool
by Jerry Richards 21 Oct '09

21 Oct '09

Can anyone recommend a good 3rd party dialplan tool that will work with Freeswitch? Best Regards, Jerry

3 2

Is background sound|music during call possible?
by Joey Carter 21 Oct '09

21 Oct '09

Hello How can I add background music that will play during call? Thanks

2 1

Stop/Restart of Freeswitch Causes Crash
by Jerry Richards 21 Oct '09

21 Oct '09

Sometimes if I stop (using "..." command) and then restart freeswitch (using "./freeswitch" command), the program will crash and return to the Linux (CentOS 5.3) prompt. I am using version 1.0.4. I just pasted the freeswitch/terminal log into the Pastebin. Best Regards, Jerry

3 4