June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

in bypass_media mode, 2833 payload type modified by freeswitch!
by qingquan luo 16 Jun '11

16 Jun '11

Hi All, I use freeswitch to bridge one incoming call to other target phone number. I use bypass_media mode. So the rtp is not go through the freeswitch, By I notice that When LegB reply 183 or 200 SDP message. Freeswitch change it 2833 telephone-event payload type. and forwarding the message to caller. This make the caller use wrong dtmf payload type to send dtmf What wrong with that? How can fix it? Any information or help is welcome. information with "sofia global siptrace on" http://pastebin.freeswitch.org/16496 Thanks Best Regards Qingquan -- Using Gmail? Please read this important notice: http://www.fsf.org/campaigns/jstrap/gmail?40922.

5 10

using fs api to originate a call and record it
by wes-fs＠499x.com 15 Jun '11

15 Jun '11

I'm extremely new to freeswitch, and anything like it, so please forgive me! I'd like to be able to use the API to originate a call and record it. So far, I have the following figured out: originate sofia/sipinterface_1/912625551212(a)141.xxx.xxx.xxx &record(/tmp/myrecording.wav) And it dials out to an external number and successfully records the conversation. But, it also records the ringing. When I was trying this through a dialplan, I found an option: <action application="set" data="media_bug_answer_req=true"/> which delayed the recording until the call was answered. Is there a way to do the same thing through the API? Thanks! Wes

3 5

failure to start a session from within lua for outbound call
by paul＠iamfine.com 15 Jun '11

15 Jun '11

Having problems getting a session created from a lua script - its the latest build on a centos system this is the lua code ----------------------- freeswitch.console_log("info", "Lua in da house!!!\n"); local session = freeswitch.Session("sofia/00.00.127.61/1000"); session:execute("playback", "/sr8k.wav"); session:hangup(); ----------------------- ***** please note i changed the ip address to protect its identity ***** ********** please help - i am tearing my hair out on this one Paul +OK freeswitch@internal> 2011-06-14 20:31:01.476406 [INFO] switch_cpp.cpp:1197 Lua in da house!!! 2011-06-14 20:31:01.476406 [DEBUG] switch_ivr_originate.c:1869 Parsing global variables 2011-06-14 20:31:01.476406 [NOTICE] switch_channel.c:833 New Channel sofia/internal/1000 [c9c5d8ec-af7f-4153-a844-5000983170d5] 2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:4392 (sofia/internal/1000) State Change CS_NEW -> CS_INIT 2011-06-14 20:31:01.476406 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK] 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000) Running State Change CS_INIT 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1000) State INIT 2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:85 sofia/internal/1000 SOFIA INIT 2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:125 (sofia/internal/1000) State Change CS_INIT -> CS_ROUTING 2011-06-14 20:31:01.476406 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK] 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1000) State INIT going to sleep 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000) Running State Change CS_ROUTING 2011-06-14 20:31:01.476406 [DEBUG] switch_channel.c:1736 (sofia/internal/1000) Callstate Change DOWN -> RINGING 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1000) State ROUTING 2011-06-14 20:31:01.476406 [DEBUG] mod_sofia.c:148 sofia/internal/1000 SOFIA ROUTING 2011-06-14 20:31:01.476406 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1000) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2011-06-14 20:31:01.476406 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK] 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1000) State ROUTING going to sleep 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000) Running State Change CS_CONSUME_MEDIA 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/1000) State CONSUME_MEDIA 2011-06-14 20:31:01.476406 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/1000) State CONSUME_MEDIA going to sleep 2011-06-14 20:31:01.476406 [DEBUG] sofia.c:4800 Channel sofia/internal/1000 entering state [calling][0] 2011-06-14 20:31:01.638509 [DEBUG] sofia.c:4800 Channel sofia/internal/1000 entering state [terminated][503] 2011-06-14 20:31:01.638509 [DEBUG] switch_channel.c:2641 (sofia/internal/1000) Callstate Change RINGING -> HANGUP 2011-06-14 20:31:01.638509 [NOTICE] sofia.c:5522 Hangup sofia/internal/1000 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-06-14 20:31:01.638509 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2011-06-14 20:31:01.638509 [ERR] switch_cpp.cpp:655 session is not initalized 2011-06-14 20:31:01.638509 [ERR] switch_cpp.cpp:617 session is not initalized 2011-06-14 20:31:01.638509 [DEBUG] switch_channel.c:2657 Send signal sofia/internal/1000 [KILL] 2011-06-14 20:31:01.638509 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK] 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000) Running State Change CS_HANGUP 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1000) State HANGUP 2011-06-14 20:31:01.638509 [DEBUG] mod_sofia.c:452 sofia/internal/1000 Overriding SIP cause 503 with 503 from the other leg 2011-06-14 20:31:01.638509 [DEBUG] mod_sofia.c:458 Channel sofia/internal/1000 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1000) State HANGUP going to sleep 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1000) State Change CS_HANGUP -> CS_REPORTING 2011-06-14 20:31:01.638509 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK] 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000) Running State Change CS_REPORTING 2011-06-14 20:31:01.638509 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1000) State REPORTING 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1000 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1000) State REPORTING going to sleep 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1000) State Change CS_REPORTING -> CS_DESTROY 2011-06-14 20:31:02.631557 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1000 [BREAK] 2011-06-14 20:31:02.631557 [DEBUG] switch_core_session.c:1290 Session 8 (sofia/internal/1000) Locked, Waiting on external entities 2011-06-14 20:31:02.631557 [NOTICE] switch_core_session.c:1308 Session 8 (sofia/internal/1000) Ended 2011-06-14 20:31:02.631557 [NOTICE] switch_core_session.c:1310 Close Channel sofia/internal/1000 [CS_DESTROY] 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/1000) Callstate Change HANGUP -> DOWN 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/1000) Running State Change CS_DESTROY 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1000) State DESTROY 2011-06-14 20:31:02.631557 [DEBUG] mod_sofia.c:363 sofia/internal/1000 SOFIA DESTROY 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1000 Standard DESTROY 2011-06-14 20:31:02.631557 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1000) State DESTROY going to sleep

2 1

invalid application lua
by Wes 15 Jun '11

15 Jun '11

I'm getting "invalid application lua" however, in modules.conf, the line for <load module="mod_lua"/> is active and not commented out. Any help is appreciated. Thanks.

4 7

FreeSWITCH Conference Call Today
by Michael Collins 15 Jun '11

15 Jun '11

Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_15 We have a light agenda but we do want to talk about a few things, in particular the FreeSWITCH Cookbook and the new mod_rtmp that was released on Monday. Talk to you soon, Michael

1 0

[Fwd: Re: Warning - SIP auth challenge]
by Henk Oegema 15 Jun '11

15 Jun '11

But the message is every 4 minutes. -------- Forwarded Message -------- From: Avi Marcus <avi(a)avimarcus.net> Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Warning - SIP auth challenge Date: Wed, 15 Jun 2011 01:38:29 +0300 It's just a notice that there's a register attempt. If it's every 30 or 60 minutes from a known extension, it's fine. It's there for fail2ban to notice excessive register attempts so it can block abuse from clients that try to register but don't submit a password to avoid setting off the warning bells. -Avi On Wed, Jun 15, 2011 at 1:33 AM, Henk Oegema <henk(a)oegema.com> wrote: What is the meaning of getting (regular) this message: .....[WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1001(a)10.0.0.101] from ip 10.0.0.12 ?? Extension 1001 is a Portech GSM gateway MV370. _______________________________________________ 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 _______________________________________________ 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

5 4

Proxy traffic security
by Antonio 15 Jun '11

15 Jun '11

Hello I have a FreeSwitch configured as a gateway, the proxy makes the user authentication and other functions. All servers have public ip's, and I have doubts to security. I had thought to put the proxy ip's in acl. Is it safe this scenario? Can it be improved? Should we also put the sip providers? Thanks. Greetings

3 6

custom ODBC timeouts
by François Delawarde 15 Jun '11

15 Jun '11

Hello everyone, I would like to manually set the timeouts for ODBC (login, query and connections), to use with DBH in lua, in order to quickly fallback to another action if something goes wrong. Those timeouts look like they are hard-coded in FS, but is there any way to force them externally (odbc.ini or similar)? Thanks, François.

1 0

Fw: caller phone hangup call after callee answer
by hala alramli 15 Jun '11

15 Jun '11

i have kamailio server with freeswitch but i face this problem .any help to explain the reason. i have a problem here that when callee answer caller phone send bye and terminate the call i will attach the track. freeswitch listen to sip_server:5068 U 2011/06/13 13:23:41.764946 caller_ip_XXXXX:1024 -> sip_server:5060 INVITE sip:callee@sip..XXXX.XXXSIP/2.0. Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-a9c2d8ca. From: <sip:909999999@sip.XXXX.XXX>;tag=4fde77c876fbda1ao0. To: <sip:callee@sip.XXXX.XXX>. Call-ID: 850573e0-95b74ce2(a)192.168.1.130. CSeq: 101 INVITE. Max-Forwards: 70. Contact: <sip:909999999@192.168.1.130:5060>. Expires: 240. User-Agent: Linksys/PAP2T-3.1.15(LS). Content-Length: 444. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 785603 785603 IN IP4 192.168.1.130. s=-. c=IN IP4 192.168.1.130. t=0 0. m=audio 16416 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 2011/06/13 13:23:41.765588 sip_server:5060 -> caller_ip_XXXXX:1024 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-a9c2d8ca;rport=1024;received=caller_ip_XXXXX. From: <sip:909999999@sip.XXXX.XXX>;tag=4fde77c876fbda1ao0. To: <sip:callee@sip.XXXX.XXX>;tag=4471d251144740103ef2d5e5631bdc66.e595. Call-ID: 850573e0-95b74ce2(a)192.168.1.130. CSeq: 101 INVITE. Proxy-Authenticate: Digest realm="sip.XXXX.XXX", nonce="4df5f4690000f03a53f01a12380d236e373437474bf441ef". Server: Kamailio (1.4.4-notls (x86_64/linux)). Content-Length: 0. . U 2011/06/13 13:23:41.822266 caller_ip_XXXXX:1024 -> sip_server:5060 ACK sip:callee@sip..XXXX.XXXSIP/2.0. Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-a9c2d8ca. From: <sip:909999999@sip.XXXX.XXX>;tag=4fde77c876fbda1ao0. To: <sip:callee@sip.XXXX.XXX>;tag=4471d251144740103ef2d5e5631bdc66.e595. Call-ID: 850573e0-95b74ce2(a)192.168.1.130. CSeq: 101 ACK. Max-Forwards: 70. Contact: <sip:909999999@192.168.1.130:5060>. User-Agent: Linksys/PAP2T-3.1.15(LS). Content-Length: 0. . U 2011/06/13 13:23:41.834779 caller_ip_XXXXX:1024 -> sip_server:5060 INVITE sip:callee@sip..XXXX.XXXSIP/2.0. Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-9f4eddee. From: <sip:909999999@sip.XXXX.XXX>;tag=4fde77c876fbda1ao0. To: <sip:callee@sip.XXXX.XXX>. Call-ID: 850573e0-95b74ce2(a)192.168.1.130. CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="909999999",realm="sip.XXXX.XXX",nonce="4df5f4690000f03a53f01a12380d236e373437474bf441 ef",uri="sip:callee@sip.XXXX.XXX",algorithm=MD5,response="82a5c2648544cf1abd616f7816a61a5f". Contact: <sip:909999999@192.168.1.130:5060>. Expires: 240. User-Agent: Linksys/PAP2T-3.1.15(LS). Content-Length: 444. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 785603 785603 IN IP4 192.168.1.130. s=-. c=IN IP4 192.168.1.130. t=0 0. m=audio 16416 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 2011/06/13 13:23:41.838496 sip_server:5060 -> caller_ip_XXXXX:1024 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-9f4eddee;rport=1024;received=caller_ip_XXXXX. From: <sip:909999999@sip.XXXX.XXX>;tag=4fde77c876fbda1ao0. To: <sip:callee@sip.XXXX.XXX>. Call-ID: 850573e0-95b74ce2(a)192.168.1.130. CSeq: 102 INVITE. Server: Kamailio (1.4.4-notls (x86_64/linux)). Content-Length: 0. . U 2011/06/13 13:23:41.843647 sip_server:5068 -> gateway_XXXX:5060 INVITE sip:122callee@gateway_XXXX SIP/2.0. Via: SIP/2.0/UDP sip_server:5068;rport;branch=z9hG4bKcZmy3ypB0jm1B. Max-Forwards: 14. From: "909999999" <sip:909999999@sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr. To: <sip:122callee@gateway_XXXX>. Call-ID: 6f179185-1052-122f-49bf-00221988c79d. CSeq: 13646046 INVITE. Contact: <sip:ufs@sip_server:5068>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11078. 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. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 478. P-PInfo: 0. Remote-Party-ID: "909999999" <sip:909999999@sip.XXXX.XXX>;screen=yes;privacy=off. . v=0. o=- 4131904911551689921 6268338771912655224 IN IP4 sip_server. s=-. c=IN IP4 sip_server. t=0 0. m=audio 19594 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=direction:active. U 2011/06/13 13:23:41.847833 gateway_XXXX:5060 -> sip_server:5068 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server:5068;received=sip_server;rport=5068;branch=z9hG4bKcZmy3ypB0jm1B. From: "909999999" <sip:909999999@sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr. To: <sip:122callee@gateway_XXXX>. Call-ID: 6f179185-1052-122f-49bf-00221988c79d. CSeq: 13646046 INVITE. Content-Length: 0. . U 2011/06/13 13:23:47.038040 gateway_XXXX:5060 -> sip_server:5068 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP sip_server:5068;received=sip_server;rport=5068;branch=z9hG4bKcZmy3ypB0jm1B. From: "909999999" <sip:909999999@sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr. To: <sip:122callee@gateway_XXXX>;tag=SDlfkl999-e4e6842982432011613141134. Call-ID: 6f179185-1052-122f-49bf-00221988c79d. CSeq: 13646046 INVITE. Server: CS2000_NGSS/8.0. Supported: 100rel. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. Contact: <sip:gateway_XXXX:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 246. . v=0. o=PVG 1307966093710 1307966093710 IN IP4 gateway_XXXX. s=-. p=+1 6135555555. c=IN IP4 gateway_XXXX. t=0 0. m=audio 36568 RTP/AVP 18 101. a=fmtp:18 annexb=no. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. U 2011/06/13 13:23:47.049924 sip_server:5060 -> caller_ip_XXXXX:1024 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.1.130:5060;rport=1024;received=caller_ip_XXXXX;branch=z9hG4bK-9f4eddee. Record-Route: <sip:sip_server;lr=on;ftag=4fde77c876fbda1ao0>. From: <sip:909999999@sip.XXXX.XXX>;tag=4fde77c876fbda1ao0. To: <sip:callee@sip.XXXX.XXX>;tag=4gv0KF120ej1c. Call-ID: 850573e0-95b74ce2(a)192.168.1.130. CSeq: 102 INVITE. Contact: <sip:mod_sofia@sip_server:5068;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11078. 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-Type: application/sdp. Content-Disposition: session. Content-Length: 248. . v=0. o=PVG 2756959439342557186 2159116678429256306 IN IP4 sip_server. s=-. p=+1 6135555555. c=IN IP4 sip_server. t=0 0. m=audio 0 RTP/AVP 99 101. a=rtpmap:99 G729/8000. a=fmtp:99 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 2011/06/13 13:23:49.770967 gateway_XXXX:5060 -> sip_server:5068 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server:5068;received=sip_server;rport=5068;branch=z9hG4bKcZmy3ypB0jm1B. From: "909999999" <sip:909999999@sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr. To: <sip:122callee@gateway_XXXX>;tag=SDlfkl999-e4e6842982432011613141134. Call-ID: 6f179185-1052-122f-49bf-00221988c79d. CSeq: 13646046 INVITE. Server: CS2000_NGSS/8.0. Supported: 100rel. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. Contact: <sip:gateway_XXXX:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 246. . v=0. o=PVG 1307966093710 1307966093710 IN IP4 gateway_XXXX. s=-. p=+1 6135555555. c=IN IP4 gateway_XXXX. t=0 0. m=audio 36568 RTP/AVP 18 101. a=fmtp:18 annexb=no. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. U 2011/06/13 13:23:49.771167 sip_server:5068 -> gateway_XXXX:5060 ACK sip:gateway_XXXX:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP sip_server:5068;rport;branch=z9hG4bKD8DQ5S7eXUamQ. Max-Forwards: 70. From: "909999999" <sip:909999999@sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr. To: <sip:122callee@gateway_XXXX>;tag=SDlfkl999-e4e6842982432011613141134. Call-ID: 6f179185-1052-122f-49bf-00221988c79d. CSeq: 13646046 ACK. Contact: <sip:ufs@sip_server:5068>. Content-Length: 0. . U 2011/06/13 13:23:49.774216 sip_server:5060 -> caller_ip_XXXXX:1024 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.130:5060;rport=1024;received=caller_ip_XXXXX;branch=z9hG4bK-9f4eddee. Record-Route: <sip:sip_server;lr=on;ftag=4fde77c876fbda1ao0>. From: <sip:909999999@sip.XXXX.XXX>;tag=4fde77c876fbda1ao0. To: <sip:callee@sip.XXXX.XXX>;tag=4gv0KF120ej1c. Call-ID: 850573e0-95b74ce2(a)192.168.1.130. CSeq: 102 INVITE. Contact: <sip:mod_sofia@sip_server:5068;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11078. 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. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 248. . v=0. o=PVG 2756959439342557186 2159116678429256306 IN IP4 sip_server. s=-. p=+1 6135555555. c=IN IP4 sip_server. t=0 0. m=audio 0 RTP/AVP 99 101. a=rtpmap:99 G729/8000. a=fmtp:99 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 2011/06/13 13:23:49.836231 caller_ip_XXXXX:1024 -> sip_server:5060 ACK sip:mod_sofia@sip_server:5068;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-d84f3d54. From: <sip:909999999@sip.XXXX.XXX>;tag=4fde77c876fbda1ao0. To: <sip:callee@sip.XXXX.XXX>;tag=4gv0KF120ej1c. Call-ID: 850573e0-95b74ce2(a)192.168.1.130. CSeq: 102 ACK. Max-Forwards: 70. Route: <sip:sip_server;lr=on;ftag=4fde77c876fbda1ao0>. Proxy-Authorization: Digest username="909999999",realm="sip.XXXX.XXX",nonce="4df5f4690000f03a53f01a12380d236e373437474bf441 ef",uri="sip:mod_sofia@sip_server:5068",algorithm=MD5,response="1dc1c1666a5d210779ab1daf84b6eb10". Contact: <sip:909999999@192.168.1.130:5060>. User-Agent: Linksys/PAP2T-3.1.15(LS). Content-Length: 0. . U 2011/06/13 13:23:49.843388 caller_ip_XXXXX:1024 -> sip_server:5060 BYE sip:mod_sofia@sip_server:5068;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-1710f527. From: <sip:909999999@sip.XXXX.XXX>;tag=4fde77c876fbda1ao0. To: <sip:callee@sip.XXXX.XXX>;tag=4gv0KF120ej1c. Call-ID: 850573e0-95b74ce2(a)192.168.1.130. CSeq: 103 BYE. Max-Forwards: 70. Route: <sip:sip_server;lr=on;ftag=4fde77c876fbda1ao0>. Proxy-Authorization: Digest username="909999999",realm="sip.XXXX.XXX",nonce="4df5f4690000f03a53f01a12380d236e373437474bf441 ef",uri="sip:mod_sofia@sip_server:5068",algorithm=MD5,response="ff113ffd721bbaef656458c39840e450". User-Agent: Linksys/PAP2T-3.1.15(LS). Content-Length: 0. . U 2011/06/13 13:23:49.845476 sip_server:5060 -> caller_ip_XXXXX:1024 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.130:5060;rport=1024;received=caller_ip_XXXXX;branch=z9hG4bK-1710f527. From: <sip:909999999@sip.XXXX.XXX>;tag=4fde77c876fbda1ao0. To: <sip:callee@sip.XXXX.XXX>;tag=4gv0KF120ej1c. Call-ID: 850573e0-95b74ce2(a)192.168.1.130. CSeq: 103 BYE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11078. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Content-Length: 0. . U 2011/06/13 13:23:49.848748 sip_server:5068 -> gateway_XXXX:5060 BYE sip:gateway_XXXX:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP sip_server:5068;rport;branch=z9hG4bKeH7F7mrjt406j. Max-Forwards: 70. From: "909999999" <sip:909999999@sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr. To: <sip:122callee@gateway_XXXX>;tag=SDlfkl999-e4e6842982432011613141134. Call-ID: 6f179185-1052-122f-49bf-00221988c79d. CSeq: 13646047 BYE. Contact: <sip:ufs@sip_server:5068>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11078. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. . U 2011/06/13 13:23:49.858691 gateway_XXXX:5060 -> sip_server:5068 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server:5068;received=sip_server;rport=5068;branch=z9hG4bKeH7F7mrjt406j. From: "909999999" <sip:909999999@sip.XXXX.XXX>;tag=5SNSNaj6XQ8Kr. To: <sip:122callee@gateway_XXXX>;tag=SDlfkl999-e4e6842982432011613141134. Call-ID: 6f179185-1052-122f-49bf-00221988c79d. CSeq: 13646047 BYE. Server: CS2000_NGSS/8.0. Supported: 100rel. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. Content-Length: 0. .

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Asterisk as gateway for FS
by William Alianto 15 Jun '11

15 Jun '11

Hi, I'm trying to setup a system using Asterisk and FreeSwitch, where Asterisk will perform as gateway for FreeSwitch. Is this setup possible? If yes, where can I find an example setup to look on to? -- Regards, William

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