Larry Alkoff
2006-Sep-01 17:42 UTC
[asterisk-users] What does 'trunk' mean in outgoing and incoming?
I'm configuring a Sipura SPA-3000 to go with my existing and working Asterisk 1.2.5 setup. The Sipura configuration files give an extension context [201] in sip.conf with the instruction "This goes into the Incoming settings for your Trunk". It also gives a extension context of [pstn-spa3k] in sip.conf with the instruction "This section goes into the Outgoing Settings for your Trunk". What does 'Incoming' and 'Outgoing' settings for your trunk mean? Where do trunks live and what are they meant to do? In my setup I have in sip.conf a [telasip-gw] context that references a context=telasip-in in extensions.conf. In extensions I have a [telasip-in] and [telasip-out] context. Which if any of these are 'trunks'? The Future of Telephony doesn't say much about trunks. Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux
Tim St. Pierre
2006-Sep-01 19:49 UTC
[asterisk-users] What does 'trunk' mean in outgoing and incoming?
Traditionally, a trunk was a group of channels interconnecting switches. On a PBX, they are the incoming and outgoing lines to the rest of the world. A trunk is any channel that can carry a call to selectable destinations, as opposed to a subscriber line that only goes to one place. In the IP world, it's more of an organizational concept. I have separate config files - sip-trunks.conf and sip-phones.conf sip-trunks is where I put config to help my machine talk to other machines. sip-phones is were the configuration for phones go. It really doesn't matter, since it's all the same file. It's just saying it that way because that's what that type of connection would be doing in a traditional telephony way of thinking. On September 1, 2006 20:42, Larry Alkoff wrote:> I'm configuring a Sipura SPA-3000 to go with my existing and working > Asterisk 1.2.5 setup. > > The Sipura configuration files give an extension context [201] in > sip.conf with the instruction "This goes into the Incoming settings for > your Trunk". > > It also gives a extension context of [pstn-spa3k] in sip.conf with the > instruction "This section goes into the Outgoing Settings for your Trunk". > > What does 'Incoming' and 'Outgoing' settings for your trunk mean? > Where do trunks live and what are they meant to do? > > In my setup I have in sip.conf a [telasip-gw] context that references a > context=telasip-in in extensions.conf. > > In extensions I have a [telasip-in] and [telasip-out] context. > > Which if any of these are 'trunks'? > > The Future of Telephony doesn't say much about trunks. > > Larry-- Tim St. Pierre IP telephony specialist sip://5101@communicatefreely.net Toronto: 647 722 6930 Toll-Free 1 888 488 6940 tim@communicatefreely.net -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 187 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060901/7655b608/attachment.pgp
Larry Alkoff
2006-Sep-12 14:24 UTC
[asterisk-users] Re: [A*UG] What does 'trunk' mean in outgoing and incoming?
Wayne Walker wrote:> Sorry, been out of touch for a while. Run asterisk -r , then > > set debug 30 > set verbose 30 > sip debug > > Now call the phone line that will cause a ring on the Sipura. Send us > the sip and normal asterisk debug out put that occurs during the ringing. > > > On Fri, Sep 01, 2006 at 07:42:08PM -0500, Larry Alkoff wrote: >> I'm configuring a Sipura SPA-3000 to go with my existing and working >> Asterisk 1.2.5 setup. >> >> The Sipura configuration files give an extension context [201] in >> sip.conf with the instruction "This goes into the Incoming settings for >> your Trunk". >> >> It also gives a extension context of [pstn-spa3k] in sip.conf with the >> instruction "This section goes into the Outgoing Settings for your Trunk". >> >> What does 'Incoming' and 'Outgoing' settings for your trunk mean? >> Where do trunks live and what are they meant to do? >> >> In my setup I have in sip.conf a [telasip-gw] context that references a >> context=telasip-in in extensions.conf. >> >> In extensions I have a [telasip-in] and [telasip-out] context. >> >> Which if any of these are 'trunks'? >> >> The Future of Telephony doesn't say much about trunks. >> >> LarryAttached is my sip.conf and extensions.conf - zip to conserve bw. Below is the result of a call with the debug instructions you sent on. It may start with the tail end of a call I aborted. For some reason, Asterisk keeps trying to destroy the call over and over. tillie*CLI> Sep 12 16:06:41 NOTICE[3271]: chan_sip.c:5242 sip_reregister: -- Re-registration for lalkoff@gw3.telasip.com REGISTER 13 headers, 0 lines REGISTER attempt 1 to lalkoff@gw3.telasip.com Reliably Transmitting (NAT) to 4.79.19.58:5060: REGISTER sip:gw3.telasip.com SIP/2.0 Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK512c64ef;rport From: <sip:lalkoff@gw3.telasip.com>;tag=as2575269c To: <sip:lalkoff@gw3.telasip.com> Call-ID: 1e2987044f31872c0c286c2b07378f0e@192.168.0.22 CSeq: 106 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="lalkoff", realm="telasip.com", algorithm=MD5, uri="sip:gw3.telasip.com", nonce="508f794e", response="87133d4c6c53b72710140cc534c05d00", opaque="" Expires: 120 Contact: <sip:5128796776@69.91.84.176> Event: registration Content-Length: 0 --- tillie*CLI> <-- SIP read from 4.79.19.58:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK512c64ef;received=69.91.84.176;rport=5060 From: <sip:lalkoff@gw3.telasip.com>;tag=as2575269c To: <sip:lalkoff@gw3.telasip.com> Call-ID: 1e2987044f31872c0c286c2b07378f0e@192.168.0.22 CSeq: 106 REGISTER User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:lalkoff@4.79.19.58> Content-Length: 0 --- (10 headers 0 lines)--- tillie*CLI> <-- SIP read from 4.79.19.58:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK512c64ef;received=69.91.84.176;rport=5060 From: <sip:lalkoff@gw3.telasip.com>;tag=as2575269c To: <sip:lalkoff@gw3.telasip.com>;tag=as53cb6b63 Call-ID: 1e2987044f31872c0c286c2b07378f0e@192.168.0.22 CSeq: 106 REGISTER User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:lalkoff@4.79.19.58> WWW-Authenticate: Digest realm="telasip.com", nonce="13fd35f9" Content-Length: 0 --- (11 headers 0 lines)--- Responding to challenge, registration to domain/host name gw3.telasip.com REGISTER 13 headers, 0 lines REGISTER attempt 2 to lalkoff@gw3.telasip.com Reliably Transmitting (NAT) to 4.79.19.58:5060: REGISTER sip:gw3.telasip.com SIP/2.0 Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK3074f777;rport From: <sip:lalkoff@gw3.telasip.com>;tag=as14bc6108 To: <sip:lalkoff@gw3.telasip.com> Call-ID: 1e2987044f31872c0c286c2b07378f0e@192.168.0.22 CSeq: 107 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="lalkoff", realm="telasip.com", algorithm=MD5, uri="sip:gw3.telasip.com", nonce="13fd35f9", response="c9d9c0d070e2a039ffa5b60e1699d72a", opaque="" Expires: 120 Contact: <sip:5128796776@69.91.84.176> Event: registration Content-Length: 0 --- tillie*CLI> <-- SIP read from 4.79.19.58:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK3074f777;received=69.91.84.176;rport=5060 From: <sip:lalkoff@gw3.telasip.com>;tag=as14bc6108 To: <sip:lalkoff@gw3.telasip.com> Call-ID: 1e2987044f31872c0c286c2b07378f0e@192.168.0.22 CSeq: 107 REGISTER User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:lalkoff@4.79.19.58> Content-Length: 0 --- (10 headers 0 lines)--- tillie*CLI> <-- SIP read from 4.79.19.58:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK3074f777;received=69.91.84.176;rport=5060 From: <sip:lalkoff@gw3.telasip.com>;tag=as14bc6108 To: <sip:lalkoff@gw3.telasip.com>;tag=as53cb6b63 Call-ID: 1e2987044f31872c0c286c2b07378f0e@192.168.0.22 CSeq: 107 REGISTER User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: <sip:5128796776@69.91.84.176>;expires=120 Date: Tue, 12 Sep 2006 21:12:01 GMT Content-Length: 0 --- (12 headers 0 lines)--- Scheduling destruction of call '1e2987044f31872c0c286c2b07378f0e@192.168.0.22' in 32000 ms Sep 12 16:06:42 NOTICE[3271]: chan_sip.c:9669 handle_response_register: Outbound Registration: Expiry for gw3.telasip.com is 120 sec (Scheduling reregistration in 105 s) tillie*CLI> <-- SIP read from 192.168.0.41:5061: INVITE sip:SIP/120@192.168.0.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.41:5061;branch=z9hG4bK-e858ec4 From: ALKOFF LAWRENCE <sip:5123017666@192.168.0.22>;tag=4eff7f365f272bf4o1 To: <sip:SIP/120@192.168.0.22> Call-ID: b3c549de-7a7a16c@192.168.0.41 CSeq: 101 INVITE Max-Forwards: 70 Contact: PSTN-IN <sip:5123017666@192.168.0.41:5061> Expires: 240 User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 420 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 89907 89907 IN IP4 192.168.0.41 s=- c=IN IP4 192.168.0.41 t=0 0 m=audio 16446 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/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=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 19 lines)--- Using INVITE request as basis request - b3c549de-7a7a16c@192.168.0.41 Sending to 192.168.0.41 : 5061 (NAT) Found peer 'pstn-spa3k' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.41:16446 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x51d (g723|ulaw|alaw|g726|g729|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for SIP/120 in home (domain 192.168.0.22) Reliably Transmitting (NAT) to 192.168.0.41:5061: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.41:5061;branch=z9hG4bK-e858ec4;received=192.168.0.41 From: ALKOFF LAWRENCE <sip:5123017666@192.168.0.22>;tag=4eff7f365f272bf4o1 To: <sip:SIP/120@192.168.0.22>;tag=as57fd29f4 Call-ID: b3c549de-7a7a16c@192.168.0.41 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:SIP/120@192.168.0.22> Content-Length: 0 --- tillie*CLI> <-- SIP read from 192.168.0.41:5061: ACK sip:SIP/120@192.168.0.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.41:5061;branch=z9hG4bK-e858ec4 From: ALKOFF LAWRENCE <sip:5123017666@192.168.0.22>;tag=4eff7f365f272bf4o1 To: <sip:SIP/120@192.168.0.22>;tag=as57fd29f4 Call-ID: b3c549de-7a7a16c@192.168.0.41 CSeq: 101 ACK Max-Forwards: 70 Contact: PSTN-IN <sip:5123017666@192.168.0.41:5061> User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 0 --- (10 headers 0 lines)--- Destroying call 'b3c549de-7a7a16c@192.168.0.41' 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.0.124:5060: OPTIONS sip:124@192.168.0.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK5d031b29;rport From: "asterisk" <sip:asterisk@192.168.0.22>;tag=as3a38c760 To: <sip:124@192.168.0.124:5060> Contact: <sip:asterisk@192.168.0.22> Call-ID: 525910d626731f3c78673c915e2af396@192.168.0.22 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 21:07:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.0.112:5060: OPTIONS sip:112@192.168.0.112:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK0bfe5594;rport From: "asterisk" <sip:asterisk@192.168.0.22>;tag=as5e5d0064 To: <sip:112@192.168.0.112:5060> Contact: <sip:asterisk@192.168.0.22> Call-ID: 430e20cd2d16c7d62aa551ac0793fab2@192.168.0.22 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 21:07:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.0.110:5060: OPTIONS sip:110@192.168.0.110:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK646607ef;rport From: "asterisk" <sip:asterisk@192.168.0.22>;tag=as3ce95fc5 To: <sip:110@192.168.0.110:5060;user=phone> Contact: <sip:asterisk@192.168.0.22> Call-ID: 3ce4853d2ab7337d7926e1634e1e0d55@192.168.0.22 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 21:07:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.0.120:5060: OPTIONS sip:120@192.168.0.120:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK089ba80a;rport From: "asterisk" <sip:asterisk@192.168.0.22>;tag=as513614bf To: <sip:120@192.168.0.120:5060> Contact: <sip:asterisk@192.168.0.22> Call-ID: 675ef5e1282dce1e53c921b668f9b775@192.168.0.22 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 21:07:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- tillie*CLI> <-- SIP read from 192.168.0.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK5d031b29;rport From: "asterisk" <sip:asterisk@192.168.0.22>;tag=as3a38c760 To: <sip:124@192.168.0.124:5060>;tag=e14e575b1eaf58a0 Call-ID: 525910d626731f3c78673c915e2af396@192.168.0.22 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.0.16 Contact: <sip:124@192.168.0.124:5060> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.0.122:5060: OPTIONS sip:122@192.168.0.122:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK244c9e3e;rport From: "asterisk" <sip:asterisk@192.168.0.22>;tag=as4bd4d56a To: <sip:122@192.168.0.122:5060;user=phone> Contact: <sip:asterisk@192.168.0.22> Call-ID: 49508eb65c016b010176919d45134a41@192.168.0.22 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 21:07:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '525910d626731f3c78673c915e2af396@192.168.0.22' tillie*CLI> <-- SIP read from 192.168.0.112:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK0bfe5594;rport From: "asterisk" <sip:asterisk@192.168.0.22>;tag=as5e5d0064 To: <sip:112@192.168.0.112:5060>;tag=eaedb459128a878d Call-ID: 430e20cd2d16c7d62aa551ac0793fab2@192.168.0.22 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.0.16 Contact: <sip:112@192.168.0.112:5060> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '430e20cd2d16c7d62aa551ac0793fab2@192.168.0.22' tillie*CLI> <-- SIP read from 192.168.0.110:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK646607ef;rport From: "asterisk" <sip:asterisk@192.168.0.22>;tag=as3ce95fc5 To: <sip:110@192.168.0.110:5060;user=phone>;tag=b2da6304d37a8958 Call-ID: 3ce4853d2ab7337d7926e1634e1e0d55@192.168.0.22 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.0.16 Contact: <sip:110@192.168.0.110:5060;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '3ce4853d2ab7337d7926e1634e1e0d55@192.168.0.22' tillie*CLI> <-- SIP read from 192.168.0.120:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK089ba80a;rport From: "asterisk" <sip:asterisk@192.168.0.22>;tag=as513614bf To: <sip:120@192.168.0.120:5060>;tag=69d7179758c7edd4 Call-ID: 675ef5e1282dce1e53c921b668f9b775@192.168.0.22 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.0.16 Contact: <sip:120@192.168.0.120:5060> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '675ef5e1282dce1e53c921b668f9b775@192.168.0.22' tillie*CLI> <-- SIP read from 192.168.0.122:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.22:5060;branch=z9hG4bK244c9e3e;rport From: "asterisk" <sip:asterisk@192.168.0.22>;tag=as4bd4d56a To: <sip:122@192.168.0.122:5060;user=phone>;tag=c8cd12ea2205d29c Call-ID: 49508eb65c016b010176919d45134a41@192.168.0.22 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.0.16 Contact: <sip:122@192.168.0.122:5060;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '49508eb65c016b010176919d45134a41@192.168.0.22' Destroying call '1e2987044f31872c0c286c2b07378f0e@192.168.0.22' 12 headers, 0 lines Reliably Transmitting (NAT) to 4.79.19.58:5060: OPTIONS sip:gw3.telasip.com SIP/2.0 Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK4feee17c;rport From: "asterisk" <sip:asterisk@69.91.84.176>;tag=as0011d202 To: <sip:gw3.telasip.com> Contact: <sip:asterisk@69.91.84.176> Call-ID: 695e4c876c421661051b5a143f542174@69.91.84.176 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 21:07:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- tillie*CLI> <-- SIP read from 4.79.19.58:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 69.91.84.176:5060;branch=z9hG4bK4feee17c From: "asterisk" <sip:asterisk@69.91.84.176>;tag=as0011d202 To: <sip:gw3.telasip.com>;tag=as0dbf1bab Call-ID: 695e4c876c421661051b5a143f542174@69.91.84.176 CSeq: 102 OPTIONS User-Agent: Telasip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:4.79.19.58> Accept: application/sdp Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '695e4c876c421661051b5a143f542174@69.91.84.176' tillie*CLI> -------------- next part -------------- A non-text attachment was scrubbed... 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