[Asterisk-Users] 2 Linphones communicating through Asterisk?

[Jon Shamash]

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" <thomas.sparr at azent.se>
To: <asterisk-users at lists.digium.com>
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:60 at 192.168.0.60
> and sip:66 at 192.168.0.66.
> If my understanding is correct they should be available on the Asterisk
> as sip:60 at 192.168.0.10 and sip:66 at 192.168.0.10.
> However, if I try to call sip:66 at 192.168.0.10 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:60 at 192.168.0.10>;tag=680816676
> To: <sip:60 at 192.168.0.10>;tag=680816676
> Call-ID: 412209507 at 192.168.0.60
> CSeq: 0 REGISTER
> Contact: <sip:60 at 192.168.0.60>
> 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:60 at 192.168.0.10>;tag=680816676
> To: <sip:60 at 192.168.0.10>;tag=680816676
> Call-ID: 412209507 at 192.168.0.60
> CSeq: 0 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:60 at 192.168.0.10>
> 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:60 at 192.168.0.10>;tag=680816676
> To: <sip:60 at 192.168.0.10>;tag=680816676
> Call-ID: 412209507 at 192.168.0.60
> CSeq: 0 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Expires: 3600
> Contact: <sip:60 at 192.168.0.10>;expires=3600
> Date: Thu, 04 Mar 2004 13:45:14 GMT
> Content-Length: 0
>
> Sip read:
> INVITE sip:66 at 192.168.0.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352
> From: <sip:60 at 192.168.0.10>;tag=4243659372;tag=4075507534
> To:  <sip:66 at 192.168.0.10>
> Call-ID: 1413225445 at 192.168.0.60
> CSeq: 20 INVITE
> Contact: <sip:60 at 192.168.0.60>
> 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:60 at 192.168.0.10>;tag=4243659372;tag=4075507534
> To: <sip:66 at 192.168.0.10>;tag=as7e281fb9
> Call-ID: 1413225445 at 192.168.0.60
> CSeq: 20 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:66 at 192.168.0.10>
> Content-Length: 0
>
>
>  to 192.168.0.60:5060
>
>
> Sip read:
> ACK sip:66 at 192.168.0.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352
> From: <sip:60 at 192.168.0.10>;tag=4243659372;tag=4075507534
> To: <sip:66 at 192.168.0.10>;tag=as7e281fb9
> Call-ID: 1413225445 at 192.168.0.60
> CSeq: 20 ACK
> Content-Length: 0
>
>
>
>