Re: [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]
Hi! Issue solved. Looks like all I was missing was one parameter: fromuser= That's interesting - could be related to this: http://lists.digium.com/pipermail/asterisk-dev/2006-November/024842.html You were probably caught be the fact that you are using extension numbers also as SIP user names for your phones (here: 3666). This is not a good thing to do, better use an alphanumeric username or the phone's MAC address etc. As for your IAX sound quality issue: I have seen that before as well, and switched to SIP (as others did). My guess is that it will probably go away if you use Asterisk 1.4 on both sides, though. SIP DEBUG on the receiving Asterisk gives you a hint which peer was found if matching is done on the IP address, the text is somethint like Found peer ... or Found no matching peer or user for w.x.y.z Two quotes from the Wiki to explain things better: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer When Asterisk receives an incoming SIP call, the SIP Channel Module 1. first tries to find a [user] section matching the caller name (From: username), 2. then tries to find a [peer] section matching the caller's IP address. 3. If no matching user or peer is found, the call is sent to the context defined in the [general] section of sip.conf. As of Asterisk 1.2, there is no reason to actually use 'user' entries any more at all; you can use 'type=peer' for everything and the behavior will be much more consistent. All configuration options supported under 'type=user' are also supported under 'type=peer'. The difference between friend and peer is the same as defining _both_ a user and peer, since that is what 'type=friend' does internally. The only benefit of type=user is when you _want_ to match on username regardless of IP the calls originate from. If the peer is registering to you, you don't need it. If they are on a fixed IP, you don't need it. 'type=peer' is _never_ matched on username for incoming calls, only matched on IP address/port number (unless you use insecure=port or higher). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]
--- On Fri, 5/14/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: You were probably caught be the fact that you are using extension numbers also as SIP user names for your phones (here: 3666). This is not a good thing to do, better use an alphanumeric username or the phone's MAC address etc. Is there more info on this? I mean, why is it bad, apart from the security implication. As for your IAX sound quality issue: I have seen that before as well, and switched to SIP (as others did). My guess is that it will probably go away if you use Asterisk 1.4 on both sides, though. It went away even with 1.2 but I needed to set trunk=no. Probably a jitter buffer issue on my system(s). SIP DEBUG on the receiving Asterisk gives you a hint which peer was found if matching is done on the IP address, the text is somethint like Found peer ... or Found no matching peer or user for w.x.y.z Tnanks for the info Philipp. I'll try to further debug my SIP messages. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]
Issue solved. Looks like all I was missing was one parameter: fromuser= Thanks for your time! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
Hi! I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls). With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious. Either a) set a secret and use that on both sides, or b) look at allowguest= and the default context and maybe the domain= settings, or c) use insecure=invite Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Either a) set a secret and use that on both sides, or b) look at allowguest= and the default context and maybe the domain= settings, or c) use insecure=invite Thanks Philipp. I'm trying option c) which is the simplest. used insecure=invite but failed with the same SIP messages. Tried also insecure=yes but the same messages show up: SIP/2.0 407 Proxy Authentication Required I had already tried a) before but did not record the SIP messages (it also failed). I haven't tried c) yet... So I'll do a) again and log the messages and then try c). Do you actually have a working SIP trunk within your LAN? If so, could you please share your settings? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both servers in
sip.conf about this servers settings.
Vardan
Vieri wrote:
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN
(no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP. I must be missing something
obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.112
context=mycontext
Asterisk server 2 (192.168.250.112) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.111
context=mycontext
I dialed from a SIP extension (4053) in server 2 (192.168.250.112) to 3666 in
server 1 (192.168.250.111) via the interboxsip SIP trunk.
The call fails and according to the SIP messages it seems to be an
authentication problem.
What am I missing?
SIP messages on 192.168.250.112 (Asterisk server 2 - transmitting call):
-- Executing [3...@from-internal:2] Dial(SIP/4053-6dea,
SIP/interboxsip/3666|300|rt) in new stack
Audio is at 192.168.250.112 port 15850
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.250.111:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:13:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 15850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called interboxsip/3666
--- SIP read from 192.168.250.111:5060 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2545a5dd
Content-Length: 0
-
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.250.111:5060:
ACK sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/interboxsip-6deb is circuit-busy
SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end):
-- SIP read from 192.168.250.112:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From: devicesip:4...@192.168.250.112;tag=as18a568d6
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 328617546726e5d430538e8061771...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:20:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
upported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 14648 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (14 headers 13 lines) ---
Using INVITE request as basis request -
328617546726e5d430538e8061771...@192.168.250.112
Sending to 192.168.250.112 : 5060 (NAT)
Reliably Transmitting (NAT) to 192.168.250.112:5060:
SIP/2.0 407 Proxy Authentication Required
Re: [asterisk-users] SIP trunk between two Asterisk servers
Vardan wrote:
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both servers in
sip.conf about this servers settings.
Vardan
Vieri wrote:
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN
(no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP. I must be missing
something obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.112
context=mycontext
Asterisk server 2 (192.168.250.112) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.111
context=mycontext
I dialed from a SIP extension (4053) in server 2 (192.168.250.112) to 3666
in server 1 (192.168.250.111) via the interboxsip SIP trunk.
The call fails and according to the SIP messages it seems to be an
authentication problem.
What am I missing?
SIP messages on 192.168.250.112 (Asterisk server 2 - transmitting call):
-- Executing [3...@from-internal:2] Dial(SIP/4053-6dea,
SIP/interboxsip/3666|300|rt) in new stack
Audio is at 192.168.250.112 port 15850
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.250.111:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:13:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 15850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called interboxsip/3666
--- SIP read from 192.168.250.111:5060 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2545a5dd
Content-Length: 0
-
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.250.111:5060:
ACK sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/interboxsip-6deb is circuit-busy
SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end):
-- SIP read from 192.168.250.112:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From: devicesip:4...@192.168.250.112;tag=as18a568d6
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 328617546726e5d430538e8061771...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:20:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
upported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 14648 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (14 headers 13 lines) ---
Using INVITE request as basis request -
328617546726e5d430538e8061771...@192.168.250.112
Sending to 192.168.250.112 : 5060 (NAT)
Reliably Transmitting (NAT) to
Re: [asterisk-users] SIP trunk between two Asterisk servers
And also please show your settings and logs (without debug) Vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: Either a) set a secret and use that on both sides, or b) look at allowguest= and the default context and maybe the domain= settings, or c) use insecure=invite Thanks Philipp. I'm trying option c) which is the simplest. used insecure=invite but failed with the same SIP messages. Tried also insecure=yes but the same messages show up: SIP/2.0 407 Proxy Authentication Required I had already tried a) before but did not record the SIP messages (it also failed). I haven't tried c) yet... So I'll do a) again and log the messages and then try c). Do you actually have a working SIP trunk within your LAN? If so, could you please share your settings? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
I have forget to write for outcall in extension
server1:
[calltoserver2]
exten = _X.,1,Noop(Call to server2)
exten = _X.,2,Dial(SIP/interboxserver2/${EXTEN})
exten = _X.,3,Hangup
server2:
[calltoserver1]
exten = _X.,1,Noop(Call to server1)
exten = _X.,2,Dial(SIP/interboxserver1/${EXTEN})
exten = _X.,3,Hangup
:)
Vardan
Vardan wrote:
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both servers in
sip.conf about this servers settings.
Vardan
Vieri wrote:
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN
(no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP. I must be missing
something obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.112
context=mycontext
Asterisk server 2 (192.168.250.112) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.111
context=mycontext
I dialed from a SIP extension (4053) in server 2 (192.168.250.112) to 3666
in server 1 (192.168.250.111) via the interboxsip SIP trunk.
The call fails and according to the SIP messages it seems to be an
authentication problem.
What am I missing?
SIP messages on 192.168.250.112 (Asterisk server 2 - transmitting call):
-- Executing [3...@from-internal:2] Dial(SIP/4053-6dea,
SIP/interboxsip/3666|300|rt) in new stack
Audio is at 192.168.250.112 port 15850
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.250.111:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:13:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 15850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called interboxsip/3666
--- SIP read from 192.168.250.111:5060 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2545a5dd
Content-Length: 0
-
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.250.111:5060:
ACK sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From: devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Contact:sip:4...@192.168.250.112
Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/interboxsip-6deb is circuit-busy
SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end):
-- SIP read from 192.168.250.112:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From: devicesip:4...@192.168.250.112;tag=as18a568d6
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID: 328617546726e5d430538e8061771...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:20:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
upported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 14648 RTP/AVP 0 8 101
Re: [asterisk-users] SIP trunk between two Asterisk servers
Hi! I'm trying option c) which is the simplest. used insecure=invite but failed with the same SIP messages. Tried also insecure=yes but the same messages show up: SIP/2.0 407 Proxy Authentication Required Then you have another entry in sip.conf that uses the same IP address. Delete that, or change the port on one of them, and adjust insecure= accordingly. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
Hi again! --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote:
I have forget to write for outcall in
extension
server1:
[calltoserver2]
exten = _X.,1,Noop(Call to server2)
exten =
_X.,2,Dial(SIP/interboxserver2/${EXTEN})
exten = _X.,3,Hangup
server2:
[calltoserver1]
exten = _X.,1,Noop(Call to server1)
exten =
_X.,2,Dial(SIP/interboxserver1/${EXTEN})
exten = _X.,3,Hangup
:)
Vardan
Vardan wrote:
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both
servers in
sip.conf about this servers settings.
Vardan
Vieri wrote:
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk
servers on the same LAN (no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP.
I must be missing something obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf
contains:
[interboxsip]
type=peer
host=192.168.250.112
context=mycontext
Asterisk server 2 (192.168.250.112) sip.conf
contains:
[interboxsip]
type=peer
host=192.168.250.111
context=mycontext
I dialed from a SIP extension (4053) in server 2
(192.168.250.112) to 3666 in server 1 (192.168.250.111) via
the interboxsip SIP trunk.
The call fails and according to the SIP messages
it seems to be an authentication problem.
What am I missing?
SIP messages on 192.168.250.112 (Asterisk server 2
- transmitting call):
-- Executing
[3...@from-internal:2] Dial(SIP/4053-6dea,
SIP/interboxsip/3666|300|rt) in new stack
Audio is at 192.168.250.112 port 15850
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to
192.168.250.111:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:13:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 15850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called
interboxsip/3666
--- SIP read from 192.168.250.111:5060
---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5,
realm=asterisk, nonce=2545a5dd
Content-Length: 0
-
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.250.111:5060:
ACK sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Contact:sip:4...@192.168.250.112
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
--
SIP/interboxsip-6deb is circuit-busy
SIP messages on 192.168.250.111 (Asterisk server 1
- receiving end):
-- SIP read from 192.168.250.112:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From:
devicesip:4...@192.168.250.112;tag=as18a568d6
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID:
328617546726e5d430538e8061771...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:20:26 GMT
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111 Contact: sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111;tag=as57a19dac Contact: sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: SIP/2.0 407 Proxy Authentication Required Then you have another entry in sip.conf that uses the same IP address. Delete that, or change the port on one of them, and adjust insecure= accordingly. asterisk1 # grep 192.168.250 sip*.conf sip.conf:host=192.168.250.112 asterisk2 # grep 192.168.250 sip*.conf sip.conf:host=192.168.250.111 So I only have 1 entry in each server's sip.conf and this entry is in interboxsip (my sample SIP trunk name). Puzzling... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
Please look in any conf file that have any relations with sip.conf.
I think you have some records.
And one also, you take this message when calling in both direction?
(server1 call server2 and server2 call server1)
Vardan
Vieri wrote:
--- On Wed, 5/12/10, Vardanhvarda...@gmail.com wrote:
I have forget to write for outcall in
extension
server1:
[calltoserver2]
exten = _X.,1,Noop(Call to server2)
exten =
_X.,2,Dial(SIP/interboxserver2/${EXTEN})
exten = _X.,3,Hangup
server2:
[calltoserver1]
exten = _X.,1,Noop(Call to server1)
exten =
_X.,2,Dial(SIP/interboxserver1/${EXTEN})
exten = _X.,3,Hangup
:)
Vardan
Vardan wrote:
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both
servers in
sip.conf about this servers settings.
Vardan
Vieri wrote:
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk
servers on the same LAN (no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP.
I must be missing something obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf
contains:
[interboxsip]
type=peer
host=192.168.250.112
context=mycontext
Asterisk server 2 (192.168.250.112) sip.conf
contains:
[interboxsip]
type=peer
host=192.168.250.111
context=mycontext
I dialed from a SIP extension (4053) in server 2
(192.168.250.112) to 3666 in server 1 (192.168.250.111) via
the interboxsip SIP trunk.
The call fails and according to the SIP messages
it seems to be an authentication problem.
What am I missing?
SIP messages on 192.168.250.112 (Asterisk server 2
- transmitting call):
-- Executing
[3...@from-internal:2] Dial(SIP/4053-6dea,
SIP/interboxsip/3666|300|rt) in new stack
Audio is at 192.168.250.112 port 15850
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to
192.168.250.111:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:13:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 15850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called
interboxsip/3666
--- SIP read from 192.168.250.111:5060
---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5,
realm=asterisk, nonce=2545a5dd
Content-Length: 0
-
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.250.111:5060:
ACK sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Contact:sip:4...@192.168.250.112
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
--
SIP/interboxsip-6deb is circuit-busy
SIP messages on 192.168.250.111 (Asterisk server 1
- receiving end):
-- SIP read from 192.168.250.112:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From:
devicesip:4...@192.168.250.112;tag=as18a568d6
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID:
328617546726e5d430538e8061771...@192.168.250.112
CSeq: 102 INVITE
Re: [asterisk-users] SIP trunk between two Asterisk servers
please show sip show users and sip show peers vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
And sip show registry Vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: please show sip show users and sip show peers SERVER 2: sip show users (trimmed to just my sip test trunk): Username Secret Accountcode Def.Context ACL NAT interboxsip mycontext No RFC3581 sip show peers (also trimmed): Name/username HostDyn Nat ACL Port Status sipprovider/01 w.x.y.zN 5060 OK (90 ms) interboxsip192.168.250.111 5060 Unmonitored 7503/7503 10.215.146.190 D N A 5060 OK (20 ms) 7502/7502 10.215.146.203 D N A 5060 OK (20 ms) 7172/7172 192.168.250.7D N A 13404OK (40 ms) 7166/7166 10.215.146.200 D N A 5060 OK (20 ms) 7165/7165 10.215.248.12D N A 5060 OK (1 ms) 7160/7160 10.215.146.182 D N A 5060 OK (20 ms) 7137/7137 192.168.250.6D N A 25967OK (10 ms) 7118/7118 192.168.250.10 D N A 14508OK (1 ms) 7117/7117 10.215.146.185 D N A 5060 OK (20 ms) 7114/7114 192.168.250.8D N A 12342OK (10 ms) 7112/7112 192.168.250.31 D N A 19829OK (10 ms) 7111/7111 192.168.250.32 D N A 35259OK (80 ms) 7109/7109 (Unspecified)D N A 0UNKNOWN 7097/7097 10.215.146.164 D N A 5060 OK (20 ms) SERVER 1: sip show users is identical. sip show peers (trimmed): Name/username HostDyn Nat ACL Port Status sipprovider/01 w.x.y.zN 5060 OK (79 ms) interboxsip192.168.250.112 5060 Unmonitored vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong?
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: And sip show registry sip show registry doesn't list anything regarding my interboxsip test trunk because I'm trying to setup a straightforward link such as this one described here (without user/password): http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/ The only sip show registry entry I have is the one for my external Internet SIP trunk, which is ok. Thanks for your time. Vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
Please change the peers name in any server. for example: server1: interboxsip1 server2: interboxsip2 Vardan Vieri wrote: --- On Wed, 5/12/10, Vardanhvarda...@gmail.com wrote: please show sip show users and sip show peers SERVER 2: sip show users (trimmed to just my sip test trunk): Username Secret Accountcode Def.Context ACL NAT interboxsip mycontext No RFC3581 sip show peers (also trimmed): Name/username HostDyn Nat ACL Port Status sipprovider/01 w.x.y.zN 5060 OK (90 ms) interboxsip192.168.250.111 5060 Unmonitored 7503/7503 10.215.146.190 D N A 5060 OK (20 ms) 7502/7502 10.215.146.203 D N A 5060 OK (20 ms) 7172/7172 192.168.250.7D N A 13404OK (40 ms) 7166/7166 10.215.146.200 D N A 5060 OK (20 ms) 7165/7165 10.215.248.12D N A 5060 OK (1 ms) 7160/7160 10.215.146.182 D N A 5060 OK (20 ms) 7137/7137 192.168.250.6D N A 25967OK (10 ms) 7118/7118 192.168.250.10 D N A 14508OK (1 ms) 7117/7117 10.215.146.185 D N A 5060 OK (20 ms) 7114/7114 192.168.250.8D N A 12342OK (10 ms) 7112/7112 192.168.250.31 D N A 19829OK (10 ms) 7111/7111 192.168.250.32 D N A 35259OK (80 ms) 7109/7109 (Unspecified)D N A 0UNKNOWN 7097/7097 10.215.146.164 D N A 5060 OK (20 ms) SERVER 1: sip show users is identical. sip show peers (trimmed): Name/username HostDyn Nat ACL Port Status sipprovider/01 w.x.y.zN 5060 OK (79 ms) interboxsip192.168.250.112 5060 Unmonitored vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] SIP trunk between two Asterisk servers
What are your allowguest= and domain= settings in the global section of sip.conf? And which version of Asterisk exactly are you using? Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: What are your allowguest= and domain= settings in the global section of sip.conf? And which version of Asterisk exactly are you using? I have no such settings defined yet. Still haven't tried to set them... Not sure what to put in domain. Anyway: # /etc/asterisk/sip.conf [general] vmexten=*97 disallow=all allow=ulaw allow=alaw context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 rtptimeout=120 rtpholdtimeout=300 pedantic=no urlencode=yes register=01:...@internet_sip_provider.com/01010101010101 regcontext=dundi-extens Server 2: Asterisk 1.4.31 Server 1: same sip.conf settings except Asterisk 1.2.40 Notice the urlencode setting which is a patch taken from: https://issues.asterisk.org/view.php?id=14652 This may be the culprit but I'm not quite sure about it. Also, I *need* this patch unless the address incomplete issue gets solved. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: Please change the peers name in any server. for example: server1: interboxsip1 server2: interboxsip2 If I understand correctly, the peer names can be identical on both servers. What counts is the host entry, I guess. But then again, my SIP trunk isn't working so I'll try out your suggestion tomorrow. Thanks, Vieri Vardan Vieri wrote: --- On Wed, 5/12/10, Vardanhvarda...@gmail.com wrote: please show sip show users and sip show peers SERVER 2: sip show users (trimmed to just my sip test trunk): Username Secret Accountcode Def.Context ACL NAT interboxsip mycontext No RFC3581 sip show peers (also trimmed): Name/username Host Dyn Nat ACL Port Status sipprovider/01 w.x.y.z N 5060 OK (90 ms) interboxsip 192.168.250.111 5060 Unmonitored 7503/7503 10.215.146.190 D N A 5060 OK (20 ms) 7502/7502 10.215.146.203 D N A 5060 OK (20 ms) 7172/7172 192.168.250.7 D N A 13404 OK (40 ms) 7166/7166 10.215.146.200 D N A 5060 OK (20 ms) 7165/7165 10.215.248.12 D N A 5060 OK (1 ms) 7160/7160 10.215.146.182 D N A 5060 OK (20 ms) 7137/7137 192.168.250.6 D N A 25967 OK (10 ms) 7118/7118 192.168.250.10 D N A 14508 OK (1 ms) 7117/7117 10.215.146.185 D N A 5060 OK (20 ms) 7114/7114 192.168.250.8 D N A 12342 OK (10 ms) 7112/7112 192.168.250.31 D N A 19829 OK (10 ms) 7111/7111 192.168.250.32 D N A 35259 OK (80 ms) 7109/7109 (Unspecified) D N A 0 UNKNOWN 7097/7097 10.215.146.164 D N A 5060 OK (20 ms) SERVER 1: sip show users is identical. sip show peers (trimmed): Name/username Host Dyn Nat ACL Port Status sipprovider/01 w.x.y.z N 5060 OK (79 ms) interboxsip 192.168.250.112 5060 Unmonitored vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP
