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Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)
by Peter P GMX 24 Aug '09

24 Aug '09

Hello, when calling from Fritzbox to a Snom Phone , sound is fine. But when calling an internal Freeswitch number (conference, mailbox) i hear a very choppy voice coming from the fritzbox side. I think it may have to do with the ptime 20msec/30msec. Example: When calling from the fritzbox to a voicemail then the annoucement from Freeswitch is choppy (too slow with interrups), but the recorded message is fine. Did anybody experience the same problem? Best regards Peter Here are some SIP messages: U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060 INVITE sip:0123456@my.domain SIP/2.0. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>. Call-ID: 3125316C8A2A13B8(a)112.xxx.xx.xxx. CSeq: 55 INVITE. Contact: <sip:02xxxxxxxxx@112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>. Max-Forwards: 70. Expires: 120. User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009). Supported: 100rel,replaces,timer. Allow-Events: telephone-event,refer. Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH. Content-Type: application/sdp. Accept: application/sdp, multipart/mixed. Accept-Encoding: identity. Content-Length: 359. . v=0. o=user 16423733 16423733 IN IP4 112.xxx.xx.xxx. s=call. c=IN IP4 112.xxx.xx.xxx. t=0 0. m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101. a=sendrecv. a=rtpmap:2 G726-32/8000. 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. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=rtcp:7079. # U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>. Call-ID: 3125316C8A2A13B8(a)112.xxx.xx.xxx. CSeq: 55 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Content-Length: 0. . # U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>;tag=7t1e8BQg5B7yK. Call-ID: 3125316C8A2A13B8(a)112.xxx.xx.xxx. CSeq: 55 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Proxy-Authenticate: Digest realm="my.domain", nonce="900b46a0-8d88-11de-a6a1-098738f35adb", algorithm=MD5, qop="auth". Content-Length: 0. . # U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060 ACK sip:0123456@my.domain SIP/2.0. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>;tag=7t1e8BQg5B7yK. Call-ID: 3125316C8A2A13B8(a)112.xxx.xx.xxx. CSeq: 55 ACK. User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009). Content-Length: 0. . # U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060 INVITE sip:0123456@my.domain SIP/2.0. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>. Call-ID: 3125316C8A2A13B8(a)112.xxx.xx.xxx. CSeq: 56 INVITE. Contact: <sip:02xxxxxxxxx@112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>. Proxy-Authorization: Digest username="02xxxxxxxxx", realm="my.domain", nonce="900b46a0-8d88-11de-a6a1-098738f35adb", uri="sip:0123456@my.domain", response="276b44e261c13bd17218adff1150f414", algorithm=MD5, cnonce="CADBE5D624516E8A", qop=auth, nc=00000001. Max-Forwards: 70. Expires: 120. User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009). Supported: 100rel,replaces,timer. Allow-Events: telephone-event,refer. Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH. Content-Type: application/sdp. Accept: application/sdp, multipart/mixed. Accept-Encoding: identity. Content-Length: 359. . v=0. o=user 16423733 16423733 IN IP4 112.xxx.xx.xxx. s=call. c=IN IP4 112.xxx.xx.xxx. t=0 0. m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101. a=sendrecv. a=rtpmap:2 G726-32/8000. 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=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=rtcp:7079. # U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>. Call-ID: 3125316C8A2A13B8(a)112.xxx.xx.xxx. CSeq: 56 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Content-Length: 0. . # U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>;tag=83t7967K2mXHF. Call-ID: 3125316C8A2A13B8(a)112.xxx.xx.xxx. CSeq: 56 INVITE. Contact: <sip:0123456@182.xxx.xx.xxx:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 249. . v=0. o=FreeSWITCH 1250746184 1250746185 IN IP4 182.xxx.xx.xxx. s=FreeSWITCH. c=IN IP4 182.xxx.xx.xxx. t=0 0. m=audio 26670 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. # U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060 ACK sip:0123456@182.xxx.xx.xxx:5060;transport=udp SIP/2.0. Via: SIP/2.0/udp 112.xxx.xx.xxx:5060;branch=z9hG4bK39E0B6E237E408FC. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>;tag=83t7967K2mXHF. Call-ID: 3125316C8A2A13B8(a)112.xxx.xx.xxx. CSeq: 56 ACK. Contact: <sip:02xxxxxxxxx@112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>. Max-Forwards: 70. User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009). Content-Length: 0. . # U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060 INVITE sip:02xxxxxxxxx@112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3 SIP/2.0. Via: SIP/2.0/UDP 182.xxx.xx.xxx;rport;branch=z9hG4bK7rpNeaN67y9ta. Max-Forwards: 70. From: <sip:0123456@my.domain>;tag=83t7967K2mXHF. To: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. Call-ID: 3125316C8A2A13B8(a)112.xxx.xx.xxx. CSeq: 119268091 INVITE. Contact: <sip:0123456@182.xxx.xx.xxx:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 249. . v=0. o=FreeSWITCH 1250746184 1250746186 IN IP4 182.xxx.xx.xxx. s=FreeSWITCH. c=IN IP4 182.xxx.xx.xxx. t=0 0. m=audio 26670 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:30.

7 14

Problem with cnam.js?
by Lars Zeb 24 Aug '09

24 Aug '09

I think there's something wrong with the script at http://wiki.freeswitch.org/wiki/Examples_cnam.js. If you use it as is, it displays "Content-type: text/html" for the effective_caller_id_name. In cnam.pl, the first two output lines are generated by: if (!$debug) {print "Content-type: text/html\n\n";} with the actual name in the third line. So I changed: fd.open("read"); buff = fd.readln(); if(buff) { logger(buff, "info"); session.setVariable("effective_caller_id_name", buff); } To: fd.open("read"); buff = fd.readAll(); if(buff[2]) { logger(buff, "info"); session.setVariable("effective_caller_id_name", buff[2]); } Or remove the print statement from cnam.pl. Sorry for the code, but the page was not editable. Lars

2 1

topology hiding leaking information in SDP
by Hound Dog 24 Aug '09

24 Aug '09

Brian, I set proxy_media to false ad now getting clean SDP without all the incoming data thank you this also means that the packets have to go all the way in and out of the codec chain ( instead of being taken in and thrown out with no processing ) , is that the case ? what is the performance hit of such change , on a modern strong machine - what is the estimated streams that FS can take ? BTW - sorry for not answering the list as I should have , I think I registered wrong somewhere so now have a new user

2 2

SIPp issues - seems FS doesn't understand ACK message
by Tihomir Culjaga 24 Aug '09

24 Aug '09

Hello, I've been with freeswittch for a while now.. and i can say it is worth developing it. anyhow i got into a strange issue... I'm tryng to see what load FS on my server can take. The Call flow is like this: SIPp FS INVITE --------> <------- 100 Trying <------- 302 Moved Temporary ACK ---------> I use a dummy dialplan for that. All custom functions i've build are disabled and i'm not using it here. Also custom modules are not loaded as well. <extension name="ServiceLookup"> <condition field="destination_number" expression="(^300030)(.*)">  <action application="log" data="INFO ######################## ServiceLookup ########################\n"/> <action application="log" data="INFO ######################## contact = '${red_contact}' ##############\n"/> <action application="log" data="INFO ######################## CallerNum = '${caller_id_number:6:16}' ##########\n"/> <action application="log" data="INFO ######################## RADIUS auth = '${authResult}' ##########\n"/> <action application="execute_extension" data="doRedirect XML public"/> </condition> </extension> <extension name="doRedirect"> <condition field="destination_number" expression="^doRedirect$"/> <condition field="${authResult}" expression="^0$|^60$"> <action application="log" data="INFO ######################## RADIUS auth OK!!!' ##########\n"/>   <action application="redirect" data=" sip:12345616094191500@pgw01.ot.hr:5060"/>   <action application="hangup" data="USER_BUSY"/> <anti-action application="redirect" data=" sip:12345616094191500@pgw01.ot.hr:5060"/> <anti-action application="log" data="INFO ######################## RADIUS auth NOK!! ##########\n"/>  <anti-action application="hangup" data="USER_BUSY"/> </condition> </extension> When i place a call from x-lite everything works fine ... x-lite sends an invite, gets SIP 302 and ACKs it correctly... FS is happy. When i place a call from SIPp i have the same scenario except FS seems not understand ACK message from SIPp and re-sends SIP 302 multiple times untill it gives up. I beleive this is due to 302 resend issue but; when i load FS with 100 CPS, i can see high CPU usage (just one thread taking most load... the rest does almost nothing) on FS. Also, starting from 40 CPS there is a big delay in receiving SIP 302 messages meaning i've sent 6000 calls and so far only for half of them got 302 response. Does anybody have a clue ? Here is a trace taken on FS for calls originated from SIPp (sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg -inf test.txt -m 1 -l 4000): freeswitch@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 16:44:26.527236: ------------------------------------------------------------------------ INVITE sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251>SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport Max-Forwards: 70 Contact: <sip:22222238515000403@10.4.4.252<sip%3A22222238515000403(a)10.4.4.252> > To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> > From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 Call-ID: 1-6962(a)10.4.4.252 CSeq: 1 INVITE Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 131 v=0 o=user1 53655765 2353687637 IN IP4 10.4.4.252 s=- c=IN IP4 10.4.4.252 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ------------------------------------------------------------------------ send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> > Call-ID: 1-6962(a)10.4.4.252 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 To: "30003016094191500" <sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >;tag=Hr4mHDUeBSNyH Call-ID: 1-6962(a)10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809: ------------------------------------------------------------------------ ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> > From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 Call-ID: 1-6962(a)10.4.4.252 CSeq: 1 ACK Contact: sip:sipp@10.4.4.252:5060 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 To: "30003016094191500" <sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >;tag=Hr4mHDUeBSNyH Call-ID: 1-6962(a)10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 To: "30003016094191500" <sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >;tag=Hr4mHDUeBSNyH Call-ID: 1-6962(a)10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 Tihomir.

2 1

topology hiding leaking information in SDP data
by Hound Dog 24 Aug '09

24 Aug '09

carriers need topology hiding , its an important feature for both security and also to hide you business partners from each other freeSwitch talks about it and also does a good job in hiding the signalling topology there is however a hole in the SDP manipulation that I am trying to plug and would love to get some help , obviously once resolved I am also happy to add documentation for all to use I was thinking that the best way would be to build the SDP message from scratch based on the incoming info , and maybe in special cases have the SDP copied over from the original message. is there a way to have FS build a clean SDP message ? see and example of Bria softphone making a call via freeswitch , note that the SDP to leg B contains original addresses and even internal ones incoming Invite message ------------------- INVITE sip:442078562101@pbx.rilcomm.com SIP/2.0 Via: SIP/2.0/UDP 82.80.130.222:26762;branch=z9hG4bK-d8754z-8fba35a96b6d949f-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:44558678567378@82.80.130.222:26762> To: <sip:442078562101@pbx.rilcomm.com> From: <sip:44558678567378@pbx.rilcomm.com>;tag=47360c64 Call-ID: YTY4NjMwMjg4MWRmODY5NDlhOWQ4MDg5MWIwN2Y3MTY. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: Bria release 2.5.4 stamp 53956 Content-Length: 325 v=0 o=- 0 2 IN IP4 82.80.130.222 s=CounterPath Bria c=IN IP4 82.80.130.222 t=0 0 m=audio 27848 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 2 : XlhJjZhm LLmd1Kyz 192.168.1.50 24514 a=alt:2 1 : GOsgLipv VjRq6zYk 192.168.1.60 24514 ------------------- message to leg B see the original IP addresses in the SDP ( anything that is not 81.89.136.231) note 82.80.130.222 which is the original address being visible in the SDP fields ------------------- INVITE sip:1001@82.80.130.222:20014 SIP/2.0 Via: SIP/2.0/UDP 81.89.136.231:5080;rport;branch=z9hG4bKBBgvK7Sap23tN Max-Forwards: 67 From: "44558678567378" <sip:44558678567378@81.89.136.231>;tag=1FUrD0t1gF55a To: <sip:1001@82.80.130.222:20014> Call-ID: 8eed1cbc-9030-11de-87e7-27e3a7b5c9da CSeq: 119414118 INVITE Contact: <sip:mod_sofia@81.89.136.231:5080> User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 331 Remote-Party-ID: "44558678567378" <sip:44558678567378@81.89.136.231>;party=calling;screen=yes;privacy=off v=0 o=- 2301954626585387485 2 IN IP4 82.80.130.222 s=CounterPath Bria c=IN IP4 81.89.136.231 t=0 0 m=audio 25974 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 2 : XlhJjZhm LLmd1Kyz 192.168.1.50 24514 a=alt:2 1 : GOsgLipv VjRq6zYk 192.168.1.60 24514 ------------------- thank you Ori

3 3

CHANNEL_CREATE and full information
by Mikhail Krivushin 24 Aug '09

24 Aug '09

I put some variable in originate command for identify created channel for my app using event socket. Is any ability to get all channel variables without uuid_show on channel_create? Can I ask FS to take me all information on channel creating? Can this be in trunk, if it cannt be get without source editing? May be we can add custom event - CHANNEL_CREATE_FULL_INFO? I just think, that is smarty to push channel information for controller in channel create. I can call api uuid_call, but in some situations, why I call api, channel destroyed. I can use bgapi result, BACKGROUND_JOB, but not in all cases. Sometimes we have situation when call initiated not from my caller app. In this case we *cant* get information about destroyed channel. Please, point me to right side - may be I need to write something module?

2 1

unable to download non-US sound files
by Juan Backson 24 Aug '09

24 Aug '09

Hi I tried to download non-US sound files, but I am getting this error: --21:50:09-- http://files.freeswitch.org/freeswitch-sounds-fr-1.0.10.tar.gz Resolving files.freeswitch.org... 69.174.57.101 Connecting to files.freeswitch.org|69.174.57.101|:80... connected. HTTP request sent, awaiting response... 404 Not Found 21:50:12 ERROR 404: Not Found. The command I used is make sounds-fr-install Any idea what's wrong? Thanks, JB

3 2

Screaming monkeys on ext 5000
by Scott Torr 24 Aug '09

24 Aug '09

Just a quick note, and I'm sure why, but screaming monkeys does not play on the the default installation. I have not looked into why, but thought I would just quickly let you know. Perhaps I have not done something? regards, sbt

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Transporting SIP over TCP
by Tzury Bar Yochay 24 Aug '09

24 Aug '09

Hi, Googeling about this shows that FS aims to support this (in fact it supports all 3: UDP/TCP/TLS). Yet I could not find the way to configure FS in order to support that. In fact, it does not work in my current install. I have TLS configured and work, but could not make TCP works thanks in advance /tzury

5 24

Media Not Heard When Bridging 2 Calls with same Gateway
by Matthew Fong 24 Aug '09

24 Aug '09

I'm trying to get FreeSWITCH to bridge two channels together through the same external gateway, but I'm having issues hearing audio. Both legs if setup independently and forwarded to 5000 (test ivr) work fine for both inbound and outbound media, but when I try to bridge them together, everything seems fine in FreeSWITCH, but neither party can hear the other speak. I'm thinking the external gateway might be having some issues because I've been able to bridge 2 channels together through the same gateway on different providers, but thought I'd also try to seek some help here. FreeSWITCH should be handling the media for both channels, so I can't figure out why if Leg A and Leg B work independently, but not if they are bridged together. Is there a setting somewhere in FS that I'm missing? Below is a ngrep of the SIP interactions if it's useful. Thanks for the help. --matt interface: eth0 (172.24.200.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060 INVITE sip:914159927717@38.98.58.148 <sip%3A914159927717(a)38.98.58.148>SIP/2.0. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg. Max-Forwards: 70. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=ZtFvjeFQmXvpp. To: <sip:914159927717@38.98.58.148 <sip%3A914159927717(a)38.98.58.148>>. Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe. CSeq: 119257811 INVITE. Contact: <sip:gw+epik.com@216.81.56.198:5080;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 293. Remote-Party-ID: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198. s=FreeSWITCH. c=IN IP4 216.81.56.198. t=0 0. m=audio 24700 RTP/AVP 0 8 3 101 13. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080 SIP/2.0 100 Trying. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=ZtFvjeFQmXvpp. To: <sip:914159927717@38.98.58.148 <sip%3A914159927717(a)38.98.58.148> >;tag=F725.2C49. Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe. CSeq: 119257811 INVITE. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg. Contact: <sip:914159927717@38.98.58.148:5060>. Content-Length: 0. . U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080 SIP/2.0 183 Session Progress. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=ZtFvjeFQmXvpp. To: <sip:914159927717@38.98.58.148 <sip%3A914159927717(a)38.98.58.148> >;tag=F725.2C49. Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe. CSeq: 119257811 INVITE. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg. Contact: <sip:914159927717@38.98.58.148:5060>. Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE. Content-Type: application/sdp. Content-Length: 227. . v=0. o=BroadSoft 23178 23178 IN IP4 10.10.10.11. s=M6 Call. c=IN IP4 38.98.58.148. t=0 0. m=audio 42554 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=rtcp:6461 IN IP4 10.10.24.50. U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080 SIP/2.0 200 OK. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=ZtFvjeFQmXvpp. To: <sip:914159927717@38.98.58.148 <sip%3A914159927717(a)38.98.58.148> >;tag=F725.2C49. Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe. CSeq: 119257811 INVITE. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg. Contact: <sip:914159927717@38.98.58.148:5060>. Session-Expires: 1800;refresher=uas. Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE. Supported: timer. Content-Type: application/sdp. Content-Length: 227. . v=0. o=BroadSoft 23178 23178 IN IP4 10.10.10.11. s=M6 Call. c=IN IP4 38.98.58.148. t=0 0. m=audio 42554 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=rtcp:6461 IN IP4 10.10.24.50. U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060 ACK sip:914159927717@38.98.58.148:5060 SIP/2.0. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc. Max-Forwards: 70. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=ZtFvjeFQmXvpp. To: <sip:914159927717@38.98.58.148 <sip%3A914159927717(a)38.98.58.148> >;tag=F725.2C49. Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe. CSeq: 119257811 ACK. Contact: <sip:gw+epik.com@216.81.56.198:5080;transport=udp>. Content-Length: 0. . U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060 INVITE sip:914154650027@38.98.58.148 <sip%3A914154650027(a)38.98.58.148>SIP/2.0. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q. Max-Forwards: 70. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=038mm9ZtH6j9H. To: <sip:914154650027@38.98.58.148 <sip%3A914154650027(a)38.98.58.148>>. Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe. CSeq: 119257815 INVITE. Contact: <sip:gw+epik.com@216.81.56.198:5080;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 293. Remote-Party-ID: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198. s=FreeSWITCH. c=IN IP4 216.81.56.198. t=0 0. m=audio 24798 RTP/AVP 0 8 3 101 13. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080 SIP/2.0 100 Trying. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=038mm9ZtH6j9H. To: <sip:914154650027@38.98.58.148 <sip%3A914154650027(a)38.98.58.148> >;tag=F72E.2D4E. Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe. CSeq: 119257815 INVITE. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q. Contact: <sip:914154650027@38.98.58.148:5060>. Content-Length: 0. . U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080 SIP/2.0 183 Session Progress. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=038mm9ZtH6j9H. To: <sip:914154650027@38.98.58.148 <sip%3A914154650027(a)38.98.58.148> >;tag=F72E.2D4E. Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe. CSeq: 119257815 INVITE. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q. Contact: <sip:914154650027@38.98.58.148:5060>. Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE. Content-Type: application/sdp. Content-Length: 225. . v=0. o=BroadSoft 2035 2035 IN IP4 10.10.10.11. s=M6 Call. c=IN IP4 38.98.58.148. t=0 0. m=audio 46520 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=rtcp:6451 IN IP4 10.10.24.50. U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080 SIP/2.0 200 OK. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=038mm9ZtH6j9H. To: <sip:914154650027@38.98.58.148 <sip%3A914154650027(a)38.98.58.148> >;tag=F72E.2D4E. Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe. CSeq: 119257815 INVITE. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q. Contact: <sip:914154650027@38.98.58.148:5060>. Session-Expires: 1800;refresher=uas. Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE. Supported: timer. Content-Type: application/sdp. Content-Length: 225. . v=0. o=BroadSoft 2035 2035 IN IP4 10.10.10.11. s=M6 Call. c=IN IP4 38.98.58.148. t=0 0. m=audio 46520 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=rtcp:6451 IN IP4 10.10.24.50. U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060 ACK sip:914154650027@38.98.58.148:5060 SIP/2.0. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK. Max-Forwards: 70. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=038mm9ZtH6j9H. To: <sip:914154650027@38.98.58.148 <sip%3A914154650027(a)38.98.58.148> >;tag=F72E.2D4E. Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe. CSeq: 119257815 ACK. Contact: <sip:gw+epik.com@216.81.56.198:5080;transport=udp>. Content-Length: 0. . U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080 BYE sip:gw+epik.com@216.81.56.198:5080 SIP/2.0. From: <sip:914159927717@38.98.58.148 <sip%3A914159927717(a)38.98.58.148> >;tag=F725.2C49. To: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=ZtFvjeFQmXvpp. Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe. CSeq: 4817 BYE. Max-Forwards: 70. Via: SIP/2.0/UDP 38.98.58.148:5060 ;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3. Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819. Contact: <sip:914159927717@38.98.58.148:5060>. Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE. Content-Length: 0. . U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 38.98.58.148:5060 ;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3. Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819. From: <sip:914159927717@38.98.58.148 <sip%3A914159927717(a)38.98.58.148> >;tag=F725.2C49. To: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=ZtFvjeFQmXvpp. Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe. CSeq: 4817 BYE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Content-Length: 0. . U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060 BYE sip:914154650027@38.98.58.148:5060 SIP/2.0. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF. Max-Forwards: 70. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=038mm9ZtH6j9H. To: <sip:914154650027@38.98.58.148 <sip%3A914154650027(a)38.98.58.148> >;tag=F72E.2D4E. Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe. CSeq: 119257816 BYE. Contact: <sip:gw+epik.com@216.81.56.198:5080;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. . U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080 SIP/2.0 200 OK. From: "FreeSWITCH" <sip:0000000000@216.81.56.198<sip%3A0000000000(a)216.81.56.198> >;tag=038mm9ZtH6j9H. To: <sip:914154650027@38.98.58.148 <sip%3A914154650027(a)38.98.58.148> >;tag=F72E.2D4E. Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe. CSeq: 119257816 BYE. Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF. Contact: <sip:914154650027@38.98.58.148:5060>. Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE. Content-Length: 0. .

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