Matthew, SIP.conf [100] username=100 secret=password type=friend host=dynamic nat=yes canreinvite=no insecure=port disallow=all allow=ulaw allow=alaw allow=g729 context=asterisk qualify=no dialplan [asterisk] exten => _X.,1,AGI(call.php) exten => h,1,AGI(hangup.php) SIP Trace: 201.xxx.xxx.xxx = SIP Softphone which originates the call xxx.xxx.xxx.xxx = Asterisk server yyy.yyy.yyy.yyy = ITSP <--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---> INVITE sip:12127773...@xxx.xxx.xxx.xxx SIP/2.0 To: <sip:12127773...@xxx.xxx.xxx.xxx> From: 100<sip:1...@xxx.xxx.xxx.xxx>;tag=7c0c4b22 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-181300058-1--d87543-;rport Call-ID: 6601fe453f41d566 CSeq: 1 INVITE Contact: <sip:1...@201.xxx.xxx.xxx:5060> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Content-Length: 228 v=0 o=- 7847157 7847631 IN IP4 201.xxx.xxx.xxx s=eyeBeam c=IN IP4 201.xxx.xxx.xxx t=0 0 m=audio 8614 RTP/AVP 0 101 a=alt:1 1 : 05A48429 0000007F 201.xxx.xxx.xxx 8614 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jun 3 13:11:27] --- (12 headers 10 lines) --- [Jun 3 13:11:27] == Using SIP RTP CoS mark 5 [Jun 3 13:11:27] Sending to 201.xxx.xxx.xxx : 5060 (no NAT) [Jun 3 13:11:27] Using INVITE request as basis request - 6601fe453f41d566 [Jun 3 13:11:27] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060 [Jun 3 13:11:27] <--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-181300058-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 From: 100<sip:1...@xxx.xxx.xxx.xxx>;tag=7c0c4b22 To: <sip:12127773...@xxx.xxx.xxx.xxx>;tag=as1999a2fb Call-ID: 6601fe453f41d566 CSeq: 1 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c9deee3" Content-Length: 0 <------------> [Jun 3 13:11:27] Scheduling destruction of SIP dialog '6601fe453f41d566' in 32000 ms (Method: INVITE) [Jun 3 13:11:27] <--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---> INVITE sip:12127773...@xxx.xxx.xxx.xxx SIP/2.0 To: <sip:12127773...@xxx.xxx.xxx.xxx> From: 100<sip:1...@xxx.xxx.xxx.xxx>;tag=7c0c4b22 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;rport Call-ID: 6601fe453f41d566 CSeq: 2 INVITE Contact: <sip:1...@201.xxx.xxx.xxx:5060> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="100",realm="asterisk",nonce="0c9deee3",uri="sip:12127773...@xxx.xxx.xxx.xxx",response="e84d9090cfc8e60f94768583218ae0ad",algorithm=MD5 Content-Length: 228 v=0 o=- 7847157 7847631 IN IP4 201.xxx.xxx.xxx s=eyeBeam c=IN IP4 201.xxx.xxx.xxx t=0 0 m=audio 8614 RTP/AVP 0 101 a=alt:1 1 : 05A48429 0000007F 201.xxx.xxx.xxx 8614 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jun 3 13:11:27] --- (13 headers 10 lines) --- [Jun 3 13:11:27] Sending to 201.xxx.xxx.xxx : 5060 (NAT) [Jun 3 13:11:27] Using INVITE request as basis request - 6601fe453f41d566 [Jun 3 13:11:27] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060 [Jun 3 13:11:27] Found RTP audio format 0 [Jun 3 13:11:27] Found RTP audio format 101 [Jun 3 13:11:27] Found audio description format telephone-event for ID 101 [Jun 3 13:11:27] Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jun 3 13:11:27] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 3 13:11:27] Peer audio RTP is at port 201.xxx.xxx.xxx:8614 [Jun 3 13:11:27] Looking for 12127773456 in asterisk (domain xxx.xxx.xxx.xxx) [Jun 3 13:11:27] list_route: hop: <sip:1...@201.xxx.xxx.xxx:5060> [Jun 3 13:11:27] <--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 From: 100<sip:1...@xxx.xxx.xxx.xxx>;tag=7c0c4b22 To: <sip:12127773...@xxx.xxx.xxx.xxx> Call-ID: 6601fe453f41d566 CSeq: 2 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:12127773...@xxx.xxx.xxx.xxx> Content-Length: 0 <------------> [Jun 3 13:11:27] -- Executing [12127773456@asterisk:1] AGI("SIP/100-000034d8", "call.php") in new stack [Jun 3 13:11:27] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php [Jun 3 13:11:28] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456) [Jun 3 13:11:28] == Using SIP RTP CoS mark 5 [Jun 3 13:11:28] Audio is at xxx.xxx.xxx.xxx port 56248 [Jun 3 13:11:28] Adding codec 0x4 (ulaw) to SDP [Jun 3 13:11:28] Adding non-codec 0x1 (telephone-event) to SDP [Jun 3 13:11:28] Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:5060: INVITE sip:12127773...@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport Max-Forwards: 70 From: "100" <sip:1...@xxx.xxx.xxx.xxx>;tag=as643c20b1 To: <sip:12127773...@yyy.yyy.yyy.yyy> Contact: <sip:1...@xxx.xxx.xxx.xxx> Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 102 INVITE User-Agent: PBX Date: Mon, 03 Jun 2013 13:11:28 IST Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 248 v=0 o=WP 1365830908 1365830908 IN IP4 xxx.xxx.xxx.xxx s=PBX c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 56248 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jun 3 13:11:28] -- Called yyy.yyy.yyy.yyy/12127773456 [Jun 3 13:11:28] <--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060 From: "100" <sip:1...@xxx.xxx.xxx.xxx>;tag=as643c20b1 To: <sip:12127773...@yyy.yyy.yyy.yyy>;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060 From: "100" <sip:1...@xxx.xxx.xxx.xxx>;tag=as643c20b1 To: <sip:12127773...@yyy.yyy.yyy.yyy>;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 102 INVITE Contact: <sip:12127773...@yyy.yyy.yyy.yyy:5060> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Content-Length: 234 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy s=SIP Media Capabilities c=IN IP4 zzz.zzz.zzz.zzz t=0 0 m=audio 21996 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 <-------------> [Jun 3 13:11:31] --- (11 headers 11 lines) --- [Jun 3 13:11:31] Found RTP audio format 0 [Jun 3 13:11:31] Found RTP audio format 101 [Jun 3 13:11:31] Found audio description format PCMU for ID 0 [Jun 3 13:11:31] Found audio description format telephone-event for ID 101 [Jun 3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jun 3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996 [Jun 3 13:11:31] -- SIP/yyy.yyy.yyy.yyy-000034d9 is making progress passing it to SIP/100-000034d8 [Jun 3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042 [Jun 3 13:11:31] Adding codec 0x4 (ulaw) to SDP [Jun 3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP [Jun 3 13:11:31] <--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 From: 100<sip:1...@xxx.xxx.xxx.xxx>;tag=7c0c4b22 To: <sip:12127773...@xxx.xxx.xxx.xxx>;tag=as654371b0 Call-ID: 6601fe453f41d566 CSeq: 2 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:12127773...@xxx.xxx.xxx.xxx> Content-Type: application/sdp Content-Length: 248 v=0 o=WP 1745765504 1745765504 IN IP4 xxx.xxx.xxx.xxx s=PBX c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 26042 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [0K[Jun 3 13:11:32] <--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060 From: "100" <sip:1...@xxx.xxx.xxx.xxx>;tag=as643c20b1 To: <sip:12127773...@yyy.yyy.yyy.yyy>;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 102 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: <sip:12127773...@yyy.yyy.yyy.yyy:5060> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Require: timer Supported: timer,replaces Session-Expires: 1800;refresher=uac Content-Length: 234 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy s=SIP Media Capabilities c=IN IP4 zzz.zzz.zzz.zzz t=0 0 m=audio 21996 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 <-------------> [Jun 3 13:11:32] --- (15 headers 11 lines) --- [Jun 3 13:11:32] list_route: hop: <sip:12127773...@yyy.yyy.yyy.yyy:5060> [Jun 3 13:11:32] set_destination: Parsing <sip:12127773...@yyy.yyy.yyy.yyy:5060> for address/port to send to [Jun 3 13:11:32] set_destination: set destination to yyy.yyy.yyy.yyy, port 5060 [Jun 3 13:11:32] Transmitting (NAT) to yyy.yyy.yyy.yyy:5060: ACK sip:12127773...@yyy.yyy.yyy.yyy:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK570164a3;rport Max-Forwards: 70 From: "100" <sip:1...@xxx.xxx.xxx.xxx>;tag=as643c20b1 To: <sip:12127773...@yyy.yyy.yyy.yyy>;tag=gK029aaa8c Contact: <sip:1...@xxx.xxx.xxx.xxx> Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 102 ACK User-Agent: PBX Content-Length: 0 [Jun 3 13:11:32] -- SIP/yyy.yyy.yyy.yyy-000034d9 answered SIP/100-000034d8 [Jun 3 13:11:32] Audio is at xxx.xxx.xxx.xxx port 26042 [Jun 3 13:11:32] Adding codec 0x4 (ulaw) to SDP [Jun 3 13:11:32] Adding non-codec 0x1 (telephone-event) to SDP [Jun 3 13:11:32] <--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 From: 100<sip:1...@xxx.xxx.xxx.xxx>;tag=7c0c4b22 To: <sip:12127773...@xxx.xxx.xxx.xxx>;tag=as654371b0 Call-ID: 6601fe453f41d566 CSeq: 2 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:12127773...@xxx.xxx.xxx.xxx> Content-Type: application/sdp Content-Length: 248 v=0 o=WP 1745765504 1745765505 IN IP4 xxx.xxx.xxx.xxx s=PBX c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 26042 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv [Jun 3 13:11:32] -- Packet2Packet bridging SIP/100-000034d8 and SIP/yyy.yyy.yyy.yyy-000034d9 [Jun 3 13:11:32] <--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---> ACK sip:12127773...@xxx.xxx.xxx.xxx SIP/2.0 To: <sip:12127773...@xxx.xxx.xxx.xxx>;tag=as654371b0 From: 100<sip:1...@xxx.xxx.xxx.xxx>;tag=7c0c4b22 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-558309054-1--d87543-;rport Call-ID: 6601fe453f41d566 CSeq: 2 ACK Contact: <sip:1...@201.xxx.xxx.xxx:5060> Max-Forwards: 70 User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="100",realm="asterisk",nonce="0c9deee3",uri="sip:12127773...@xxx.xxx.xxx.xxx",response="e84d9090cfc8e60f94768583218ae0ad",algorithm=MD5 Content-Length: 0 <-------------> [Jun 3 13:11:32] --- (11 headers 0 lines) --- [Jun 3 13:11:32] Really destroying SIP dialog '2d60af2732e232cf3dec876f601c38d9@127.0.0.1' Method: REGISTER [Jun 3 13:11:34] <--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---> BYE sip:12127773...@xxx.xxx.xxx.xxx SIP/2.0 To: <sip:12127773...@xxx.xxx.xxx.xxx>;tag=as654371b0 From: 100<sip:1...@xxx.xxx.xxx.xxx>;tag=7c0c4b22 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-845749512-1--d87543-;rport Call-ID: 6601fe453f41d566 CSeq: 3 BYE Contact: <sip:1...@201.xxx.xxx.xxx:5060> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="100",realm="asterisk",nonce="0c9deee3",uri="sip:12127773...@xxx.xxx.xxx.xxx",response="ae82a7f181072735f72c23f2c63020e7",algorithm=MD5 Content-Length: 0 [Jun 3 13:11:35] --- (12 headers 0 lines) --- [Jun 3 13:11:35] Sending to 201.xxx.xxx.xxx : 5060 (NAT) [Jun 3 13:11:35] <--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-845749512-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 From: 100<sip:1...@xxx.xxx.xxx.xxx>;tag=7c0c4b22 To: <sip:12127773...@xxx.xxx.xxx.xxx>;tag=as654371b0 Call-ID: 6601fe453f41d566 CSeq: 3 BYE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 [Jun 3 13:11:35] -- Executing [h@asterisk:1] AGI("SIP/100-000034d8", "hangup.php") in new stack [Jun 3 13:11:35] -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.php [Jun 3 13:11:35] -- <SIP/100-000034d8>AGI Script hangup.php completed, returning 0 [Jun 3 13:11:35] Scheduling destruction of SIP dialog '07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE) [Jun 3 13:11:35] set_destination: Parsing <sip:12127773...@yyy.yyy.yyy.yyy:5060> for address/port to send to [Jun 3 13:11:35] set_destination: set destination to yyy.yyy.yyy.yyy, port 5060 [Jun 3 13:11:35] Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:5060: BYE sip:12127773...@yyy.yyy.yyy.yyy:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK78ec518c;rport Max-Forwards: 70 From: "100" <sip:1...@xxx.xxx.xxx.xxx>;tag=as643c20b1 To: <sip:12127773...@yyy.yyy.yyy.yyy>;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 103 BYE User-Agent: PBX X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 [Jun 3 13:11:35] -- <SIP/100-000034d8>AGI Script call.php completed, returning -1 [Jun 3 13:11:35] <--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK78ec518c;rport=5060 From: "100" <sip:1...@xxx.xxx.xxx.xxx>;tag=as643c20b1 To: <sip:12127773...@yyy.yyy.yyy.yyy>;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 103 BYE Content-Length: 0 Regards, Kamlesh
> Date: Fri, 31 May 2013 08:50:38 -0500 > From: mr...@imminc.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] G.729 codec in pass-thru mode > > Kamlesh Kumar wrote: > > > > Yes that's correct, when I use u-law call works fine. > > > > In case of g729, I enabled sip debug with 'sip set debug on' and captured > > all > > the sip traces and got whatever I posted in last email. There was no other > > call on the system when I captured sip trace. Please suggest further > > proceedings. > > > Kamlesh, > > Please provide a SIP trace (sip set debug on) of a successful u-law call. I'm > especially interested in the dialog between the Asterisk server and the ITSP > in > this scenario. > > Also include the relevant sections of sip.conf and the dialplan. > > Regards, > > Matthew Roth > InterMedia Marketing Solutions > Software Engineer and Systems Developer > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users