[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: ;tag=680816676
To: ;tag=680816676
Call-ID: [EMAIL PROTECTED]
CSeq: 0 REGISTER
Contact:
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: ;tag=680816676
To: ;tag=680816676
Call-ID: [EMAIL PROTECTED]
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
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: ;tag=680816676
To: ;tag=680816676
Call-ID: [EMAIL PROTECTED]
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: ;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: ;tag=4243659372;tag=4075507534
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 20 INVITE
Contact:
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: ;tag=4243659372;tag=4075507534
To: ;tag=as7e281fb9
Call-ID: [EMAIL PROTECTED]
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
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: ;tag=4243659372;tag=4075507534
To: ;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 global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ;
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: ;tag=680816676 > To: ;tag=680816676 > Call-ID: [EMAIL PROTECTED] > CSeq: 0 REGISTER > Contact: > 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: ;tag=680816676 > To: ;tag=680816676 > Call-ID: [EMAIL PROTECTED] > CSeq: 0 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > 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: ;tag=680816676 > To: ;tag=680816676 > Call-ID: [EMAIL PROTECTED] > CSeq: 0 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Expires: 3600 > Contact: ;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: ;tag=4243659372;tag=4075507534 > To: > Call-ID: [EMAIL PROTECTED] > CSeq: 20 INVITE > Contact: > 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: ;tag=4243659372;tag=4075507534 > To: ;tag=as7e281fb9 > Call-ID: [EMAIL PROTECTED] > CSeq: 20 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > 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: ;tag=4243659372;tag=4075507534 > To: ;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 or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: ;tag=680816676 > > To: ;tag=680816676 > > Call-ID: [EMAIL PROTECTED] > > CSeq: 0 REGISTER > > Contact: > > 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: ;tag=680816676 > > To: ;tag=680816676 > > Call-ID: [EMAIL PROTECTED] > > CSeq: 0 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > > Contact: > > 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: ;tag=680816676 > > To: ;tag=680816676 > > Call-ID: [EMAIL PROTECTED] > > CSeq: 0 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > > Expires: 3600 > > Contact: ;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: ;tag=4243659372;tag=4075507534 > > To: > > Call-ID: [EMAIL PROTECTED] > > CSeq: 20 INVITE > > Contact: > > 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 > > F
