January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

VoIP SIP SDK with Cisco
by thomasirvin 23 Jan '12

23 Jan '12

Hi everyone, I have a small company and would like to use VoIP service with my Cisco System. Currently I am testing the SDK of Ozeki and it works fine with Cisco Unified CM (http://www.voip-sip-sdk.com/p_42-how-to-setup-ozeki-voip-sip-sdk-with-cisco…) so far. Has anyone experienced anything with this solution after a longer test period? Thanks in advance. Thomas -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/VoIP-SIP-SDK-with-Cisco-tp721… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

G.722 negotiation issue with AVM Fritzbox
by Roland Hänel 23 Jan '12

23 Jan '12

Hi, It seems that FreeSwitch cannot negotiate G.722 with an AVM Fritzbox device (*the* most popular VoIP box/router in Germany) because of what seems to be a concatenation of several bugs. First of all, the FritzBox sends this SDP in the INVITE: INVITE sip:+4921173062190950@192.168.232.164:6060;transport=udp;gw=duro SIP/2.0. [...] v=0. o=HuaweiSoftX3000 8451448 8451448 IN IP4 213.148.136.222. s=Sip Call. c=IN IP4 213.148.136.222. t=0 0. m=audio 28800 RTP/AVP 9 8 0 102 100 99 97. *a=rtpmap:9 G722/16000.* a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:102 G726-32/8000. a=rtpmap:100 G726-40/8000. a=rtpmap:99 G726-24/8000. a=rtpmap:97 iLBC/8000. a=ptime:30. a=fmtp:97 mode=30. which is wrong since RFC 3551, Section 4.5.2 clearly states that the RTP clock is 8kHz. We could all blame it on AVM here, but when taking a look at the "200 OK" from FreeSwitch, that's really not what should happen: SIP/2.0 200 OK. [...] v=0. o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164. s=FreeSWITCH. c=IN IP4 192.168.232.164. t=0 0. m=audio *0* RTP/AVP 96. *a=rtpmap:96 G722/8000.* So FreeSwitch announces G.722/8000 (correct), but as RTP/AVP ID 96 (9 was requested) and with *UDP Port 0*. The FritzBox BYE's the call when it receives that. What's even more funny is that in the FreeSwitch debugging, it says about the local SDP: [...] 2012-01-20 17:01:04.344716 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/external/+4921653258007(a)qsc.de] 192.168.232.164 port 23418 -> 213.148.136.222 port 28800 codec: 9 ms: 20 2012-01-20 17:01:04.344716 [DEBUG] switch_rtp.c:1659 Starting timer [soft] 160 bytes per 20ms 2012-01-20 17:01:04.344716 [DEBUG] mod_sofia.c:750 Local SDP sofia/external/+4921653258007(a)qsc.de: v=0 o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164 s=FreeSWITCH c=IN IP4 192.168.232.164 t=0 0 *m=audio 23418 RTP/AVP 9* *a=rtpmap:9 G722/8000* a=silenceSupp:off - - - - a=ptime:20 a=sendrecv So there, everything is correct. Does anyone have an idea about this one? Actually, I have taken a look at the source but I was not able to find the point where the UDP port gets lost... The full debugging trace and packet capture ist attached. Greetings, Roland # U 2012/01/20 17:01:04.323112 213.148.136.222:5060 -> 192.168.232.164:6060 INVITE sip:+4921173062190950@192.168.232.164:6060;transport=udp;gw=duro SIP/2.0. Via: SIP/2.0/UDP 213.148.136.222:5060;branch=z9hG4bK7u3e5920b061csoit5d0.1. Call-ID: SD24vi901-73c7d5e35e297c6dd1cbccbf5ab175ee-l65h8l3. To: <sip:+4921173062190950@qsc.de>. From: "+4921653258007" <sip:+4921653258007@qsc.de >;tag=SD24vi901-88lt6slg-CC-35. CSeq: 1 INVITE. Max-Forwards: 65. P-Asserted-Identity: <sip:+4921653258007@qsc.de>. Contact: <sip:021653258007@213.148.136.222:5060;transport=udp>. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. User-Agent: Huawei SoftX3000 V300R010. Supported: 100rel. Content-Length: 352. Content-Type: application/sdp. P-Called-Party-ID: sip:+4921173062190950@qsc.de. X-ORIGINAL-DDI-URI: sip:+4921173062190950@qsc.de. X-CID: mrg5ml6g71ggh5tty7yh7hc5osy5os8y@SoftX3000. . v=0. o=HuaweiSoftX3000 8451448 8451448 IN IP4 213.148.136.222. s=Sip Call. c=IN IP4 213.148.136.222. t=0 0. m=audio 28800 RTP/AVP 9 8 0 102 100 99 97. a=rtpmap:9 G722/16000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:102 G726-32/8000. a=rtpmap:100 G726-40/8000. a=rtpmap:99 G726-24/8000. a=rtpmap:97 iLBC/8000. a=ptime:30. a=fmtp:97 mode=30. # U 2012/01/20 17:01:04.323965 192.168.232.164:6060 -> 213.148.136.222:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 213.148.136.222:5060;branch=z9hG4bK7u3e5920b061csoit5d0.1. From: "+4921653258007" <sip:+4921653258007@qsc.de >;tag=SD24vi901-88lt6slg-CC-35. To: <sip:+4921173062190950@qsc.de>. Call-ID: SD24vi901-73c7d5e35e297c6dd1cbccbf5ab175ee-l65h8l3. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-cfa926d 2012-01-10 17-33-40 -0600. Content-Length: 0. . # U 2012/01/20 17:01:04.354937 192.168.232.164:6060 -> 213.148.136.222:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 213.148.136.222:5060;branch=z9hG4bK7u3e5920b061csoit5d0.1. From: "+4921653258007" <sip:+4921653258007@qsc.de >;tag=SD24vi901-88lt6slg-CC-35. To: <sip:+4921173062190950@qsc.de>;tag=ejS9yr5v3a6Qj. Call-ID: SD24vi901-73c7d5e35e297c6dd1cbccbf5ab175ee-l65h8l3. CSeq: 1 INVITE. Contact: <sip:+4921173062190950@192.168.232.164:6060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-cfa926d 2012-01-10 17-33-40 -0600. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 156. P-Asserted-Identity: "+4921173062190950" <sip:+4921173062190950@qsc.de>. . v=0. o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164. s=FreeSWITCH. c=IN IP4 192.168.232.164. t=0 0. m=audio 0 RTP/AVP 96. a=rtpmap:96 G722/8000. ------------------------ 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_NEW 2012-01-20 17:01:04.324734 [DEBUG] sofia.c:5494 Channel sofia/external/+ 4921653258007(a)qsc.de entering state [received][100] 2012-01-20 17:01:04.324734 [DEBUG] sofia.c:5505 Remote SDP: v=0 o=HuaweiSoftX3000 8451448 8451448 IN IP4 213.148.136.222 s=Sip Call c=IN IP4 213.148.136.222 t=0 0 m=audio 28800 RTP/AVP 9 8 0 102 100 99 97 a=rtpmap:9 G722/16000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:100 G726-40/8000 a=rtpmap:99 G726-24/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=ptime:30 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:380 (sofia/external/+4921653258007(a)qsc.de) State NEW 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G722:9:16000:30:64000]/[G722:9:8000:20:64000] 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:4814 Bah HUMBUG! Sticking with G722@8000h@20i 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:2919 Set Codec sofia/external/+4921653258007(a)qsc.de G722/8000 20 ms 160 samples 64000 bits 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:4930 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2012-01-20 17:01:04.324734 [DEBUG] sofia.c:5717 (sofia/external/+ 4921653258007(a)qsc.de) State Change CS_NEW -> CS_INIT 2012-01-20 17:01:04.324734 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_INIT 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:401 (sofia/external/+4921653258007(a)qsc.de) State INIT 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:85 sofia/external/+ 4921653258007(a)qsc.de SOFIA INIT 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:125 (sofia/external/+ 4921653258007(a)qsc.de) State Change CS_INIT -> CS_ROUTING 2012-01-20 17:01:04.324734 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:401 (sofia/external/+4921653258007(a)qsc.de) State INIT going to sleep 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_ROUTING 2012-01-20 17:01:04.324734 [DEBUG] switch_channel.c:1890 (sofia/external/+ 4921653258007(a)qsc.de) Callstate Change DOWN -> RINGING 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:410 (sofia/external/+4921653258007(a)qsc.de) State ROUTING 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:148 sofia/external/+ 4921653258007(a)qsc.de SOFIA ROUTING 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:104 sofia/external/+4921653258007(a)qsc.de Standard ROUTING 2012-01-20 17:01:04.324734 [INFO] mod_dialplan_xml.c:481 Processing +4921653258007 <+4921653258007>->+4921173062190950 in context default Dialplan: sofia/external/+4921653258007(a)qsc.de parsing [default->default] continue=false Dialplan: sofia/external/+4921653258007(a)qsc.de Absolute Condition [default] Dialplan: sofia/external/+4921653258007(a)qsc.de Action park() 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:154 (sofia/external/+4921653258007(a)qsc.de) State Change CS_ROUTING -> CS_EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:410 (sofia/external/+4921653258007(a)qsc.de) State ROUTING going to sleep 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:417 (sofia/external/+4921653258007(a)qsc.de) State EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:241 sofia/external/+ 4921653258007(a)qsc.de SOFIA EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:192 sofia/external/+4921653258007(a)qsc.de Standard EXECUTE EXECUTE sofia/external/+4921653258007(a)qsc.de park() 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.344716 [DEBUG] switch_ivr.c:591 sofia/external/+ 4921653258007(a)qsc.de Command Execute answer() EXECUTE sofia/external/+4921653258007(a)qsc.de answer() 2012-01-20 17:01:04.344716 [INFO] switch_nat.c:590 NAT port mapping disabled 2012-01-20 17:01:04.344716 [INFO] switch_nat.c:590 NAT port mapping disabled 2012-01-20 17:01:04.344716 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/external/+4921653258007(a)qsc.de] 192.168.232.164 port 23418 -> 213.148.136.222 port 28800 codec: 9 ms: 20 2012-01-20 17:01:04.344716 [DEBUG] switch_rtp.c:1659 Starting timer [soft] 160 bytes per 20ms 2012-01-20 17:01:04.344716 [DEBUG] mod_sofia.c:750 Local SDP sofia/external/+4921653258007(a)qsc.de: v=0 o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164 s=FreeSWITCH c=IN IP4 192.168.232.164 t=0 0 m=audio 23418 RTP/AVP 9 a=rtpmap:9 G722/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:729 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.344716 [DEBUG] switch_channel.c:3194 (sofia/external/+ 4921653258007(a)qsc.de) Callstate Change RINGING -> ACTIVE 2012-01-20 17:01:04.344716 [NOTICE] mod_dptools.c:1135 Channel [sofia/external/+4921653258007(a)qsc.de] has been answered 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.344716 [DEBUG] sofia.c:5494 Channel sofia/external/+ 4921653258007(a)qsc.de entering state [completed][200] 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.364714 [DEBUG] switch_ivr.c:591 sofia/external/+ 4921653258007(a)qsc.de Command Execute playback(silence_stream://100) EXECUTE sofia/external/+4921653258007(a)qsc.de playback(silence_stream://100) 2012-01-20 17:01:04.364714 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16@16000hz 1 channels 20ms 2012-01-20 17:01:04.464734 [DEBUG] switch_ivr_play_say.c:1678 done playing file silence_stream://100 2012-01-20 17:01:04.464734 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.484715 [DEBUG] switch_ivr.c:591 sofia/external/+ 4921653258007(a)qsc.de Command Execute sleep(900) EXECUTE sofia/external/+4921653258007(a)qsc.de sleep(900) 2012-01-20 17:01:04.624743 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.624743 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.624743 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.644728 [DEBUG] sofia.c:5494 Channel sofia/external/+ 4921653258007(a)qsc.de entering state [ready][200] 2012-01-20 17:01:04.664735 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_channel.c:2852 (sofia/external/+ 4921653258007(a)qsc.de) Callstate Change ACTIVE -> HANGUP 2012-01-20 17:01:04.684730 [NOTICE] sofia.c:623 Hangup sofia/external/+ 4921653258007(a)qsc.de [CS_EXECUTE] [NORMAL_CLEARING] 2012-01-20 17:01:04.684730 [DEBUG] switch_channel.c:2875 Send signal sofia/external/+4921653258007(a)qsc.de [KILL] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:2285 sofia/external/+4921653258007(a)qsc.de skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:2285 sofia/external/+4921653258007(a)qsc.de skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:417 (sofia/external/+4921653258007(a)qsc.de) State EXECUTE going to sleep 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_HANGUP 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:602 (sofia/external/+4921653258007(a)qsc.de) State HANGUP 2012-01-20 17:01:04.684730 [DEBUG] mod_sofia.c:463 sofia/external/+ 4921653258007(a)qsc.de Overriding SIP cause 480 with 200 from the other leg 2012-01-20 17:01:04.684730 [DEBUG] mod_sofia.c:469 Channel sofia/external/+ 4921653258007(a)qsc.de hanging up, cause: NORMAL_CLEARING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:47 sofia/external/+4921653258007(a)qsc.de Standard HANGUP, cause: NORMAL_CLEARING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:602 (sofia/external/+4921653258007(a)qsc.de) State HANGUP going to sleep 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:393 (sofia/external/+4921653258007(a)qsc.de) State Change CS_HANGUP -> CS_REPORTING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_REPORTING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:662 (sofia/external/+4921653258007(a)qsc.de) State REPORTING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:79 sofia/external/+4921653258007(a)qsc.de Standard REPORTING, cause: NORMAL_CLEARING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:662 (sofia/external/+4921653258007(a)qsc.de) State REPORTING going to sleep 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:387 (sofia/external/+4921653258007(a)qsc.de) State Change CS_REPORTING -> CS_DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1380 Session 3 (sofia/external/+4921653258007(a)qsc.de) Locked, Waiting on external entities 2012-01-20 17:01:04.684730 [NOTICE] switch_core_session.c:1398 Session 3 (sofia/external/+4921653258007(a)qsc.de) Ended 2012-01-20 17:01:04.684730 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/+4921653258007(a)qsc.de [CS_DESTROY] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:491 (sofia/external/+4921653258007(a)qsc.de) Callstate Change HANGUP -> DOWN 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:494 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:504 (sofia/external/+4921653258007(a)qsc.de) State DESTROY 2012-01-20 17:01:04.684730 [DEBUG] mod_sofia.c:374 sofia/external/+ 4921653258007(a)qsc.de SOFIA DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:86 sofia/external/+4921653258007(a)qsc.de Standard DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:504 (sofia/external/+4921653258007(a)qsc.de) State DESTROY going to sleep

4 8

upnp issue I think
by Henry Huang 23 Jan '12

23 Jan '12

I am running Asterisk on my internal network. And I had port forwarding of 5060 from public IP to my Asterisk's internal IP address. I have recently installed a few FreeSWITCH from git on remote locations. And the weird thing is whenever I am trying to register to those FreeSWITCH servers with a softphone on my laptop, the returning SIP message end up routed to the Asterisk instead of my laptop since my softphone default SIP port is 5060. But the same wouldn't happen if I am registering to remote Asterisk servers. The return SIP message will routed correctly back to my softphone. So in summary, I can register to remote Asterisk server even when I've already forwarded port 5060 to my internal Asterisk server, but I can't register to remote FreeSWITCH servers. Does anyone know the reason behind this? Henry

1 0

Cordless IP Phone
by ocset 22 Jan '12

22 Jan '12

Hi I have had great success using Yealink phones with Freeswitch and a customer has asked me for a cordless phone for their office. I want to ensure that they can keep as much functionality with a cordless phone as they have with the Yealink T28. I am looking at the Siemens C610IP phone but don't know how well it plays with Freeswitch. Can someone please shed some light on the Siemens phone or any alternative that you have successfully implemented. Thanks in advance

11 12

Re: [Freeswitch-users] announce count to user only entering conference - Parse Error (Michael Collins)
by Rodney 22 Jan '12

22 Jan '12

MC, thank you for adding this macro. it seems to work fine. for my current purposes i just created silent wavs until i need the complete voice instructions. ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, January 21, 2012 9:32 AM Subject: FreeSWITCH-users Digest, Vol 67, Issue 195 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: mod_rtmp + Akamai stream push (Anthony Minessale) 2. Re: announce count to user only entering conference - Parse Error (Michael Collins) 3. Re: G.722 negotiation issue with AVM Fritzbox (Brian West) I don't recall any initiative being done to facilitate what you describe but its open source code so nothing's impossible. On Thu, Jan 19, 2012 at 4:08 PM, Jock McKechnie <jock.mckechnie(a)gmail.com> wrote: Greetings wise(r) FreeSWITCH users; > >We've recently started using FreeSWITCH and mod_rtmp to do RTMP -> SIP >media conversion (audio only) and have been utterly delighted by how >easy and (so far) reliable this has been to both set up and run on a >daily basis. One of my superiors has gotten all excited and wonders if >we can use the same mechanism to do a stream push to Akamai for >content delivery amongst many thousands of RTMP clients (think radio >broadcast). > >It appears that the FreeSWITCH mod_rtmp module is _not_ capable of >this - it can send a fresh RTMP stream to a registered client, but it >cannot arbitrarily start pushing RTMP to an Akamai EntryPoint. Am I >correct in this assertion, or am I misunderstanding the documentation >(and google hits) available? > >My thanks to all; > > - Jock > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 I would use a phrase macro for this. In fact, this seems like such an obvious thing to do that I added a phrase macro to conf/lang/en/ivr/sounds.xml. Update to the latest latest like two seconds ago git and you will have it available. (You'll need to copy it over from the source directory to your install directory.) The new macro is called other_callers_in_conf. In your example the extension would be this: <extension name="room1"> <condition field="destination_number" expression="^301$"> <action application="set" data="conference_user_list=|"/> <action application="set" data="conf_count=${conference 301-127.0.0.1 list count}"/> <action application="log" data="INFO ${conf_count}"/>  <action application="playback" data="phrase:other_callers_in_conf:${conf_count}"/> <action application="conference" data="301-127.0.0.1@default"/> <action application="transfer" data="300 xml default" /> </condition> </extension> The phrase macro does all the work. NOTE: I need to record a few new phrase files and then tweak the macro. In the meantime I encourage you to look at the macro that I added and tailor it to your needs. -MC On Thu, Jan 19, 2012 at 12:45 PM, Rodney <notlikeme75(a)yahoo.com> wrote: Okay, I finally figured out how to do this hopefully since I know what conference they are entering and all seems to work accept when the room is empty. I am putting phrases before and after the say to make it more understandable to "some people" what the number means. > > >scenario: > > >When I enter a room with people the system says "there are" say count "other callers" because it is a count without them in the room yet, unlike the normal caller control to call the announce extension when they are already in the room. > > >this works fine when there is at least 1 other caller in the room. > > >but when there isn't another caller and the room is empty. it says "there are" say count is a parse error so no digit read (need it to say zero) then it say "other callers" then says my normal you are in the conference alone message which for now i have changed to say "zero" so it says sort of backwards "there are" "other callers' "zero" > > >ERR] mod_say.c:130 Parse Error! > > > > Could someone please help me understand why this Parse error could be happening on zero count. I would like to fix it. thank you. > > > > ><extension name="room1"> > <condition field="destination_number" expression="^301$"> ><action application="set" data="conference_user_list=|"/> >  > <action application="set" data="conf_count=${conference 301-127.0.0.1 list count}"/> > <action application="log" data="INFO ${conf_count}"/> > <action application="say" data="en number pronounced ${conf_count}"/> > <action application="playback" data="C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/othercallers.wav"/> > ><action application="conference" data="301-127.0.0.1@default"/> > <action application="transfer" data="300 xml default" /> > </condition></extension> > > > > > > > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > > What is your codec selection set to in FreeSWITCH and what rev of FreeSWITCH are you running? On Jan 20, 2012, at 4:09 PM, Roland Hänel wrote: OK you're right, I read the RfC and in fact a zero port means that the >stream is rejected. However the whole "negotiation" is still broken. If FS >rejects the incorrect G.722, OK, but what about all the other (perfectly >valid) codecs in the request (PCMA, PCMU)? > >Because FS only rejects G.722, the call is doomed to fail, even though it >could work nicely if FS just selected one of the acceptable offers. > >Roland -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian(a)freeswitch.org 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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Privacy on shared lines
by Darcy 21 Jan '12

21 Jan '12

Is there a setting to set lines to private on shared lines in the freeswitch I am using snom phones and have the shared lines working except that I would like to automatically default to private lines versus allowing barge in. Darcy Primrose

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MySQL type of read_codec and write_codec in CDR's.
by Henrik Aagaard Sørensen 21 Jan '12

21 Jan '12

I've seen the SQL script, cdr.sql, in the src-directory which should match the CDR's. But it is missing the read_codec and write_codec. I guess it should be VARCHAR's in MySQL, but can anyone verify that and what is the longest possible string (VARCHAR(80)?)?

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announce count to user only entering conference - Parse Error
by Rodney 21 Jan '12

21 Jan '12

Okay, I finally figured out how to do this hopefully since I know what conference they are entering and all seems to work accept when the room is empty. I am putting phrases before and after the say to make it more understandable to "some people" what the number means. scenario: When I enter a room with people the system says "there are" say count "other callers" because it is a count without them in the room yet, unlike the normal caller control to call the announce extension when they are already in the room. this works fine when there is at least 1 other caller in the room. but when there isn't another caller and the room is empty. it says "there are" say count is a parse error so no digit read (need it to say zero) then it say "other callers" then says my normal you are in the conference alone message which for now i have changed to say "zero" so it says sort of backwards "there are" "other callers' "zero" ERR] mod_say.c:130 Parse Error! Could someone please help me understand why this Parse error could be happening on zero count. I would like to fix it. thank you. <extension name="room1"> <condition field="destination_number" expression="^301$"> <action application="set" data="conference_user_list=|"/>  <action application="set" data="conf_count=${conference 301-127.0.0.1 list count}"/> <action application="log" data="INFO ${conf_count}"/> <action application="say" data="en number pronounced ${conf_count}"/> <action application="playback" data="C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/othercallers.wav"/>  <action application="conference" data="301-127.0.0.1@default"/> <action application="transfer" data="300 xml default" /> </condition></extension>

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mod_rtmp + Akamai stream push
by Jock McKechnie 20 Jan '12

20 Jan '12

Greetings wise(r) FreeSWITCH users; We've recently started using FreeSWITCH and mod_rtmp to do RTMP -> SIP media conversion (audio only) and have been utterly delighted by how easy and (so far) reliable this has been to both set up and run on a daily basis. One of my superiors has gotten all excited and wonders if we can use the same mechanism to do a stream push to Akamai for content delivery amongst many thousands of RTMP clients (think radio broadcast). It appears that the FreeSWITCH mod_rtmp module is _not_ capable of this - it can send a fresh RTMP stream to a registered client, but it cannot arbitrarily start pushing RTMP to an Akamai EntryPoint. Am I correct in this assertion, or am I misunderstanding the documentation (and google hits) available? My thanks to all; - Jock

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RTMP from non-flash/flex code
by jahnunya＠gmail.com 20 Jan '12

20 Jan '12

Hi. I'd like to ask if anybody succeeded to connect streams from RTMP session to a call, for example SIP, not using flex/Flash. I need a way to connect a SIP call to voice streams available from RTMP server (Red5). Which side initiates the call doesn't matter to me. I tried the flex client and it works fine: a call from Flash/flex could initiate a SIP call in FreeSWITCH. However, I need to connect streams from RTMP server (Red5). So far, every attempt to do the same failed: I can connect to FreeSWITCH via RTMP, of course, and make a call (by calling the "makeCall" method) but the streams cannot be connected. I tried the other way round: from SIP to RTMP, but it seems like the module was not meant to work this way, as it complaints about the session id/uuid. Unfortunately, I am not a C/C++ developer, so reading the code doesn't help me. Any solution or explanation of the RTMP module will be highly appreciated. Thanks in advance

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