[Asterisk-Users] 2 Linphones communicating through Asterisk?
Hi all, I'm experimenting with the following setup: An Asterisk server at 192.168.0.10. 2 Linphones at 192.168.0.60 and 192.168.0.66. The Linphones register themselves at the Asterisk as sip:[EMAIL PROTECTED] and sip:[EMAIL PROTECTED] If my understanding is correct they should be available on the Asterisk as sip:[EMAIL PROTECTED] and sip:[EMAIL PROTECTED] However, if I try to call sip:[EMAIL PROTECTED] from the other Linphone the Asterisk debug says: Looking for 66 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Are there any merciful soul on this list who can point me in the rigth direction? If your answer are RTFM, please tell me which FM to R. Asterisk sip debug follow below. Also attaching config files for Asterisk and Linphone I have messed with. All others are from make samples in asterisk. versions: Linphone 0.12.1, libosip 0.9.7, asterisk 2 days old from cvs. Regards Thomas Sip read: REGISTER sip:192.168.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 From: sip:[EMAIL PROTECTED];tag=680816676 To: sip:[EMAIL PROTECTED];tag=680816676 Call-ID: [EMAIL PROTECTED] CSeq: 0 REGISTER Contact: sip:[EMAIL PROTECTED] max-forwards: 10 expires: 3600 user-agent: oSIP/Linphone-0.12.1 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.60 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 From: sip:[EMAIL PROTECTED];tag=680816676 To: sip:[EMAIL PROTECTED];tag=680816676 Call-ID: [EMAIL PROTECTED] CSeq: 0 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.60:5060 -- Registered SIP '60' at 192.168.0.60 port 5060 expires 3600 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 From: sip:[EMAIL PROTECTED];tag=680816676 To: sip:[EMAIL PROTECTED];tag=680816676 Call-ID: [EMAIL PROTECTED] CSeq: 0 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED];expires=3600 Date: Thu, 04 Mar 2004 13:45:14 GMT Content-Length: 0 Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352 From: sip:[EMAIL PROTECTED];tag=4243659372;tag=4075507534 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE Contact: sip:[EMAIL PROTECTED] max-forwards: 10 user-agent: oSIP/Linphone-0.12.1 Content-Type: application/sdp Content-Length: 367 v=0 o=60 123456 654321 IN IP4 192.168.0.60 s=A conversation c=IN IP4 192.168.0.60 t=0 0 m=audio 7078 RTP/AVP 0 8 3 110 111 115 101 b=AS:110 20 b=AS:111 28 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:110 speex/8000/1 a=rtpmap:111 speex/16000/1 a=rtpmap:115 1015/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 11 headers, 16 lines Using latest request as basis request Sending to 192.168.0.60 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format PCMA Found description format GSM Found description format speex Found description format speex Found description format 1015 Found description format telephone-event Capabilities: us - 12, them - 526/0, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 66 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352 From: sip:[EMAIL PROTECTED];tag=4243659372;tag=4075507534 To: sip:[EMAIL PROTECTED];tag=as7e281fb9 Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.60:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352 From: sip:[EMAIL PROTECTED];tag=4243659372;tag=4075507534 To: sip:[EMAIL PROTECTED];tag=as7e281fb9 Call-ID: [EMAIL PROTECTED] CSeq: 20 ACK Content-Length: 0 ; ; Static extension configuration files, used by ; the pbx_config module. ; ; The General category is for certain variables. ; [general] ; ; If static is set to no, or omitted, then the pbx_config will rewrite ; this file when extensions are modified. Remember that all comments ; made in the file will be lost when that happens. ; ; XXX Not yet implemented XXX ; static=yes ; ; if static=yes and writeprotect=no, you can save dialplan by ; CLI command 'save dialplan' too ; writeprotect=no ; You can include other config files, use the #include command (without the ';') ; Note that this is different from the include command that includes contexts within ; other contexts. The #include command works in all asterisk configuration files. ;#include filename.conf ; The Globals category contains
Re: [Asterisk-Users] 2 Linphones communicating through Asterisk?
Hi... Being very new to A* myself I understand your fustrations with the manuals :) It looks like you've made a typo in your extensions.conf quote [sip] extern = 66,1,Dial(SIP/66) extern = 61,1,Dial(SIP/61) extern = 60,1,Dial(SIP/60) it should be exten = 66,1,Dial(SIP/66) Hope that helps Jnn - Original Message - From: Thomas Sparr [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 1:46 PM Subject: [Asterisk-Users] 2 Linphones communicating through Asterisk? Hi all, I'm experimenting with the following setup: An Asterisk server at 192.168.0.10. 2 Linphones at 192.168.0.60 and 192.168.0.66. The Linphones register themselves at the Asterisk as sip:[EMAIL PROTECTED] and sip:[EMAIL PROTECTED] If my understanding is correct they should be available on the Asterisk as sip:[EMAIL PROTECTED] and sip:[EMAIL PROTECTED] However, if I try to call sip:[EMAIL PROTECTED] from the other Linphone the Asterisk debug says: Looking for 66 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Are there any merciful soul on this list who can point me in the rigth direction? If your answer are RTFM, please tell me which FM to R. Asterisk sip debug follow below. Also attaching config files for Asterisk and Linphone I have messed with. All others are from make samples in asterisk. versions: Linphone 0.12.1, libosip 0.9.7, asterisk 2 days old from cvs. Regards Thomas Sip read: REGISTER sip:192.168.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 From: sip:[EMAIL PROTECTED];tag=680816676 To: sip:[EMAIL PROTECTED];tag=680816676 Call-ID: [EMAIL PROTECTED] CSeq: 0 REGISTER Contact: sip:[EMAIL PROTECTED] max-forwards: 10 expires: 3600 user-agent: oSIP/Linphone-0.12.1 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.60 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 From: sip:[EMAIL PROTECTED];tag=680816676 To: sip:[EMAIL PROTECTED];tag=680816676 Call-ID: [EMAIL PROTECTED] CSeq: 0 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.60:5060 -- Registered SIP '60' at 192.168.0.60 port 5060 expires 3600 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 From: sip:[EMAIL PROTECTED];tag=680816676 To: sip:[EMAIL PROTECTED];tag=680816676 Call-ID: [EMAIL PROTECTED] CSeq: 0 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED];expires=3600 Date: Thu, 04 Mar 2004 13:45:14 GMT Content-Length: 0 Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352 From: sip:[EMAIL PROTECTED];tag=4243659372;tag=4075507534 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE Contact: sip:[EMAIL PROTECTED] max-forwards: 10 user-agent: oSIP/Linphone-0.12.1 Content-Type: application/sdp Content-Length: 367 v=0 o=60 123456 654321 IN IP4 192.168.0.60 s=A conversation c=IN IP4 192.168.0.60 t=0 0 m=audio 7078 RTP/AVP 0 8 3 110 111 115 101 b=AS:110 20 b=AS:111 28 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:110 speex/8000/1 a=rtpmap:111 speex/16000/1 a=rtpmap:115 1015/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 11 headers, 16 lines Using latest request as basis request Sending to 192.168.0.60 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format PCMA Found description format GSM Found description format speex Found description format speex Found description format 1015 Found description format telephone-event Capabilities: us - 12, them - 526/0, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 66 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352 From: sip:[EMAIL PROTECTED];tag=4243659372;tag=4075507534 To: sip:[EMAIL PROTECTED];tag=as7e281fb9 Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.60:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352 From: sip:[EMAIL PROTECTED];tag=4243659372;tag=4075507534 To: sip:[EMAIL PROTECTED];tag=as7e281fb9 Call-ID: [EMAIL PROTECTED] CSeq: 20 ACK Content-Length: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
Re: [Asterisk-Users] 2 Linphones communicating through Asterisk?
On Thu, Mar 04, 2004 at 02:06:52PM -, Jon Shamash wrote: [snip] it should be exten = 66,1,Dial(SIP/66) Incidentally, is there a difference between = and =, or are both allowed? Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 Linphones communicating through Asterisk?
Both are allowed but for readability = is used on objects. Maxime - Original Message - From: Tor Houghton [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 10:19 AM Subject: Re: [Asterisk-Users] 2 Linphones communicating through Asterisk? On Thu, Mar 04, 2004 at 02:06:52PM -, Jon Shamash wrote: [snip] it should be exten = 66,1,Dial(SIP/66) Incidentally, is there a difference between = and =, or are both allowed? Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 Linphones communicating through Asterisk? Solved! Thanks Jon!
On Thu, 2004-03-04 at 15:06, Jon Shamash wrote: quote It looks like you've made a typo in your extensions.conf Doh! What a silly mistake. Yeah, it works now. Thank you very much! Regards Thomas Hi... Being very new to A* myself I understand your fustrations with the manuals :) It looks like you've made a typo in your extensions.conf quote [sip] extern = 66,1,Dial(SIP/66) extern = 61,1,Dial(SIP/61) extern = 60,1,Dial(SIP/60) it should be exten = 66,1,Dial(SIP/66) Hope that helps Jnn - Original Message - From: Thomas Sparr [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 1:46 PM Subject: [Asterisk-Users] 2 Linphones communicating through Asterisk? Hi all, I'm experimenting with the following setup: An Asterisk server at 192.168.0.10. 2 Linphones at 192.168.0.60 and 192.168.0.66. The Linphones register themselves at the Asterisk as sip:[EMAIL PROTECTED] and sip:[EMAIL PROTECTED] If my understanding is correct they should be available on the Asterisk as sip:[EMAIL PROTECTED] and sip:[EMAIL PROTECTED] However, if I try to call sip:[EMAIL PROTECTED] from the other Linphone the Asterisk debug says: Looking for 66 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Are there any merciful soul on this list who can point me in the rigth direction? If your answer are RTFM, please tell me which FM to R. Asterisk sip debug follow below. Also attaching config files for Asterisk and Linphone I have messed with. All others are from make samples in asterisk. versions: Linphone 0.12.1, libosip 0.9.7, asterisk 2 days old from cvs. Regards Thomas Sip read: REGISTER sip:192.168.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 From: sip:[EMAIL PROTECTED];tag=680816676 To: sip:[EMAIL PROTECTED];tag=680816676 Call-ID: [EMAIL PROTECTED] CSeq: 0 REGISTER Contact: sip:[EMAIL PROTECTED] max-forwards: 10 expires: 3600 user-agent: oSIP/Linphone-0.12.1 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.60 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 From: sip:[EMAIL PROTECTED];tag=680816676 To: sip:[EMAIL PROTECTED];tag=680816676 Call-ID: [EMAIL PROTECTED] CSeq: 0 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.60:5060 -- Registered SIP '60' at 192.168.0.60 port 5060 expires 3600 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 From: sip:[EMAIL PROTECTED];tag=680816676 To: sip:[EMAIL PROTECTED];tag=680816676 Call-ID: [EMAIL PROTECTED] CSeq: 0 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED];expires=3600 Date: Thu, 04 Mar 2004 13:45:14 GMT Content-Length: 0 Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352 From: sip:[EMAIL PROTECTED];tag=4243659372;tag=4075507534 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE Contact: sip:[EMAIL PROTECTED] max-forwards: 10 user-agent: oSIP/Linphone-0.12.1 Content-Type: application/sdp Content-Length: 367 v=0 o=60 123456 654321 IN IP4 192.168.0.60 s=A conversation c=IN IP4 192.168.0.60 t=0 0 m=audio 7078 RTP/AVP 0 8 3 110 111 115 101 b=AS:110 20 b=AS:111 28 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:110 speex/8000/1 a=rtpmap:111 speex/16000/1 a=rtpmap:115 1015/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 11 headers, 16 lines Using latest request as basis request Sending to 192.168.0.60 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format PCMA Found description format GSM Found description format speex Found description format speex Found description format 1015 Found description format telephone-event Capabilities: us - 12, them - 526/0, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 66 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352 From: sip:[EMAIL PROTECTED];tag=4243659372;tag=4075507534 To: sip:[EMAIL PROTECTED];tag=as7e281fb9 Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.60:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
