[Asterisk-Users] 2 Linphones communicating through Asterisk? Solved! Thanks Jon!

[Thomas Sparr]

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" <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
> >
> >
> >
> >
> 
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