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Asterisk 1.6 Integration / SIP Trunk


Hector Bravo
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Hi:

 

I'm trying to integrate pbxnsip 4 with an Asterisk 1.6 pbx using a SIP trunk (gateway).

 

Incoming calls are coming through, but outgoing calls are not working.

 

Messages in pbxnsip logs are: 403 forbidden. Asterisk accepts username/password configured in the trunk,

but tries to authenticate the pbxnsip extension that is trying to call an asterisk extension also.

 

The only way when this have worked is when:

- trunk uses 9001 (for example) as username

- pbxnsip extension originating calls is 9001 (same as trunk's username)

 

This scenario works just for one extension. I need all extensions can dial asterisk extensions and vice-versa.

 

Thanks for any help!

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  • 2 months later...
  • 1 month later...

Where:

4168885555 is the external number to call.

64.34.222.111 is the PBXNSIP pbx

79.10.172.171 is my local IP

 

 

what I don't see is the attempt to see where it connects to the asterisk server.

 

 

[9] 20110126113603: SIP Rx udp:79.10.172.171:46140:
INVITE sip:4168885555@64.34.222.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:46140;branch=z9hG4bK-d8754z-6a011e61de02ae31-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:40@79.10.172.171:46140>
To: "4168885555"<sip:4168885555@64.34.222.111>
From: "40"<sip:40@64.34.222.111>;tag=8b5ecf62
Call-ID: ZTJhOWM4ZjI1ZjA3ZjE5OTY5Njc2MzQ0NmUxNmY1NDk.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 265

v=0
o=- 1 2 IN IP4 192.168.1.110
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.110
t=0 0
m=audio 1586 RTP/AVP 107 0 8 101
a=alt:1 1 : N7XPQWhc NjvjTWRo 192.168.1.110 1586
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

[9] 20110126113603: UDP: Opening socket on 0.0.0.0:61238
[9] 20110126113603: UDP: Opening socket on 0.0.0.0:61239
[9] 20110126113603: UDP: Opening socket on [::]:61238
[9] 20110126113603: UDP: Opening socket on [::]:61239
[8] 20110126113603: Could not find a trunk (98 trunks)
[8] 20110126113603: Using outbound proxy sip:79.10.172.171:46140;transport=udp because UDP packet source did not match the via header
[9] 20110126113603: Resolve 57573222: udp 79.10.172.171 46140
[9] 20110126113603: SIP Tx udp:79.10.172.171:46140:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.110:46140;branch=z9hG4bK-d8754z-6a011e61de02ae31-1---d8754z-;rport=46140;received=79.10.172.171
From: "40" <sip:40@64.34.222.111>;tag=8b5ecf62
To: "4168885555" <sip:4168885555@64.34.222.111>;tag=7cdd5e20d0
Call-ID: ZTJhOWM4ZjI1ZjA3ZjE5OTY5Njc2MzQ0NmUxNmY1NDk.
CSeq: 1 INVITE
Content-Length: 0


[9] 20110126113603: Resolve 57573223: udp 79.10.172.171 46140
[9] 20110126113603: SIP Tx udp:79.10.172.171:46140:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 192.168.1.110:46140;branch=z9hG4bK-d8754z-6a011e61de02ae31-1---d8754z-;rport=46140;received=79.10.172.171
From: "40" <sip:40@64.34.222.111>;tag=8b5ecf62
To: "4168885555" <sip:4168885555@64.34.222.111>;tag=7cdd5e20d0
Call-ID: ZTJhOWM4ZjI1ZjA3ZjE5OTY5Njc2MzQ0NmUxNmY1NDk.
CSeq: 1 INVITE
User-Agent: Shanticom/3.4.0.3201
WWW-Authenticate: Digest realm="64.34.222.111",nonce="19be2cdcd227ea0e1467de7350f15c06",domain="sip:4168885555@64.34.222.111",algorithm=MD5
Content-Length: 0

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Whow 98 trunks is impressive! But that's not the problem here.

 

Also, it seems you are using STUN on the X-lite. Don't do that, this is just another source of trouble (symmetrical NAT). The PBX SBC fixes the problem without the STUN hack. But thats also not the problem here!

 

What is a problem is that obviously the X-lite does not try again to send the INVITE. Are you sure you have a passwort set there? And the right one?

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Whow 98 trunks is impressive! But that's not the problem here.

 

Also, it seems you are using STUN on the X-lite. Don't do that, this is just another source of trouble (symmetrical NAT). The PBX SBC fixes the problem without the STUN hack. But thats also not the problem here!

 

What is a problem is that obviously the X-lite does not try again to send the INVITE. Are you sure you have a passwort set there? And the right one?

 

I also tried it with a Snom 360... if that matters with the exact same results. I'm using a gateway with no user name or password associated with a global trunk.

 

This is the context in Asterisk for outgoing:

type=user

host=64.34.222.110

context=from-pstn

 

I have quite a few gateway trunks in asterisk, and they all work, but it seems to just absolutely fail with pbxnsip. BTW i'm running version 3.4 debian

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Well, the PBX does not treat the incoming INVITE above as a trunk call. It cannot find it in the 98 trunks and therefore it challenges it. So the snom also does not respond with another INVITE?! This would be very strange. Blacklisting was not in version 3, we can exclude that.

 

So from a PBX perspective, the PBX says: Okay, extension you want to call a number? Authenticate first, please. And that's the end of the conversation...

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This is what happens when I use a domain instead of the localhost.

 

Where:

69.39.221.107 - asterisk pbx

69.39.221.108 - pbxnsip

domain.pbxnsip.com - domain name (valid!)

4168885555 - number trying to dial

71.19.171.195 - phone external ip

192.168.1.110 - phone internal ip

 

`[7] 2011/01/26 15:45:21:	SIP Rx udp:71.19.171.195:54864:
INVITE sip:4168885555@domain.pbxnsip.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-5066863afe2ea563-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:101@192.168.1.110:54864>
To: "4168885555"<sip:4168885555@domain.pbxnsip.com>
From: "101"<sip:101@domain.pbxnsip.com>;tag=a0040960
Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 216

v=0
o=- 0 2 IN IP4 192.168.1.110
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.110
t=0 0
m=audio 50634 RTP/AVP 107 0 8 101
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
[8] 2011/01/26 15:45:21:	Could not find a trunk (98 trunks)
[8] 2011/01/26 15:45:21:	Using outbound proxy sip:71.19.171.195:54864;transport=udp because UDP packet source did not match the via header
[7] 2011/01/26 15:45:21:	SIP Tx udp:71.19.171.195:54864:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-5066863afe2ea563-1---d8754z-;rport=54864;received=71.19.171.195
From: "101" <sip:101@domain.pbxnsip.com>;tag=a0040960
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f
Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.
CSeq: 1 INVITE
Content-Length: 0

[7] 2011/01/26 15:45:21:	SIP Tx udp:71.19.171.195:54864:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-5066863afe2ea563-1---d8754z-;rport=54864;received=71.19.171.195
From: "101" <sip:101@domain.pbxnsip.com>;tag=a0040960
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f
Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.
CSeq: 1 INVITE
User-Agent: Shanticom-PBX/3.4.0.3201
WWW-Authenticate: Digest realm="domain.pbxnsip.com",nonce="2b07b5d9db75c91fe4a1361672d5bb3d",domain="sip:4168885555@domain.pbxnsip.com",algorithm=MD5
Content-Length: 0

[7] 2011/01/26 15:45:21:	SIP Rx udp:71.19.171.195:54864:
ACK sip:4168885555@domain.pbxnsip.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-5066863afe2ea563-1---d8754z-;rport
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f
From: "101"<sip:101@domain.pbxnsip.com>;tag=a0040960
Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.
CSeq: 1 ACK
Content-Length: 0

[7] 2011/01/26 15:45:21:	SIP Rx udp:71.19.171.195:54864:
INVITE sip:4168885555@domain.pbxnsip.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-0926993c2a5b1d0b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:101@192.168.1.110:54864>
To: "4168885555"<sip:4168885555@domain.pbxnsip.com>
From: "101"<sip:101@domain.pbxnsip.com>;tag=a0040960
Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username="101",realm="domain.pbxnsip.com",nonce="2b07b5d9db75c91fe4a1361672d5bb3d",uri="sip:4168885555@domain.pbxnsip.com",response="afa5cada40725938b15d1899a0f8be23",algorithm=MD5
Content-Length: 216

v=0
o=- 0 2 IN IP4 192.168.1.110
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.110
t=0 0
m=audio 50634 RTP/AVP 107 0 8 101
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
[8] 2011/01/26 15:45:21:	Tagging request with existing tag
[6] 2011/01/26 15:45:21:	Sending RTP for NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.#d202343f2f to 192.168.1.110:50634
[7] 2011/01/26 15:45:21:	SIP Tx udp:71.19.171.195:54864:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-0926993c2a5b1d0b-1---d8754z-;rport=54864;received=71.19.171.195
From: "101" <sip:101@domain.pbxnsip.com>;tag=a0040960
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f
Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.
CSeq: 2 INVITE
Content-Length: 0

[7] 2011/01/26 15:45:21:	SIP Tx udp:69.39.221.107:5060:
INVITE sip:14168885555@69.39.221.107;user=phone SIP/2.0
Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-bfb7d342eef2c1b067d371c1b04acb80;rport
From: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;tag=1266699933
To: <sip:14168885555@69.39.221.107;user=phone>
Call-ID: 634192b0@pbx
CSeq: 11026 INVITE
Max-Forwards: 70
Contact: <sip:6384858569@69.39.221.108:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: Shanticom-PBX/3.4.0.3201
Remote-Party-ID: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;party=calling;screen=yes
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 2124256490 2124256490 IN IP4 69.39.221.108
s=-
c=IN IP4 69.39.221.108
t=0 0
m=audio 48642 RTP/AVP 0 18 101
a=rtpmap:0 pcmu/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[8] 2011/01/26 15:45:21:	Play audio_moh/noise.wav
[7] 2011/01/26 15:45:21:	SIP Rx udp:69.39.221.107:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-bfb7d342eef2c1b067d371c1b04acb80;received=69.39.221.108;rport=5060
From: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;tag=1266699933
To: <sip:14168885555@69.39.221.107;user=phone>;tag=as2c68f520
Call-ID: 634192b0@pbx
CSeq: 11026 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7440bca6"
Content-Length: 0

[7] 2011/01/26 15:45:21:	Call 634192b0@pbx#1266699933: Clear last INVITE
[7] 2011/01/26 15:45:21:	SIP Tx udp:69.39.221.107:5060:
ACK sip:14168885555@69.39.221.107;user=phone SIP/2.0
Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-bfb7d342eef2c1b067d371c1b04acb80;rport
From: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;tag=1266699933
To: <sip:14168885555@69.39.221.107;user=phone>;tag=as2c68f520
Call-ID: 634192b0@pbx
CSeq: 11026 ACK
Max-Forwards: 70
Contact: <sip:6384858569@69.39.221.108:5060;transport=udp>
Remote-Party-ID: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;party=calling;screen=yes
Content-Length: 0

[5] 2011/01/26 15:45:21:	INVITE Response 401 Unauthorized: Terminate 634192b0@pbx
[7] 2011/01/26 15:45:21:	Other Ports: 26
[7] 2011/01/26 15:45:21:	Call Port: 13ccde2c-b65f537a@localhost#aa6c08a0f1
[7] 2011/01/26 15:45:21:	Call Port: 154d3f0c@pbx#1008184428
[7] 2011/01/26 15:45:21:	Call Port: 3c26992c6726-eu3cfqmr3aaa#d9f059f594
[7] 2011/01/26 15:45:21:	Call Port: 3c27bb5785a3-844ug2wf59xq#b1d600805f
[7] 2011/01/26 15:45:21:	Call Port: 3c27dbc54ecc-8a0txagwtu0c#36f3df207b
[7] 2011/01/26 15:45:21:	Call Port: 3c2fcaf65979-9e1r50rqa8om#bb2b24f3c5
[7] 2011/01/26 15:45:21:	Call Port: 3c32acca4ec5-07rrezqxi0lj#af0a49a366
[7] 2011/01/26 15:45:21:	Call Port: 3c3621cf732d-2bgalyw4iq1a#cde4b05959
[7] 2011/01/26 15:45:21:	Call Port: 3c3e5313785b-2yqj8rn1fmsx#655803ba5b
[7] 2011/01/26 15:45:21:	Call Port: 3c786c9e47bb-3ihpx3ca835r#9e907d2e63
[7] 2011/01/26 15:45:21:	Call Port: 3c92cf90a4a6-u9pjbehd2t50#a0a42bee81
[7] 2011/01/26 15:45:21:	Call Port: 4169e4ea@pbx#1720057541
[7] 2011/01/26 15:45:21:	Call Port: 47678ca033aa336e69f6d1bf7e94783e@69.39.221.107#dc142f7561
[7] 2011/01/26 15:45:21:	Call Port: 5dadb756@pbx#2108676965
[7] 2011/01/26 15:45:21:	Call Port: 643386be@pbx#896059230
[7] 2011/01/26 15:45:21:	Call Port: 6e80920f6f6840490ab2826670af2f07@69.39.221.107#d5697e696f
[7] 2011/01/26 15:45:21:	Call Port: 74e7e54c38acf7c8602305190862a751@69.39.221.107#eecc166ad8
[7] 2011/01/26 15:45:21:	Call Port: 7c6e218f@pbx#887543378
[7] 2011/01/26 15:45:21:	Call Port: 8014479f@pbx#1820803417
[7] 2011/01/26 15:45:21:	Call Port: 86e63a85@pbx#1919082329
[7] 2011/01/26 15:45:21:	Call Port: 91ec0f63@pbx#280178641
[7] 2011/01/26 15:45:21:	Call Port: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.#d202343f2f
[7] 2011/01/26 15:45:21:	Call Port: a0f3dd41@pbx#182301236
[7] 2011/01/26 15:45:21:	Call Port: c1f21b1eeff01a3d#241ad7d39c
[7] 2011/01/26 15:45:21:	Call Port: caf298fa@pbx#290526904
[7] 2011/01/26 15:45:21:	Call Port: ffea8091@pbx#965068692
[6] 2011/01/26 15:45:21:	Send codec pcmu/8000
[7] 2011/01/26 15:45:21:	SIP Tx udp:71.19.171.195:54864:
SIP/2.0 183 Ringing
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-0926993c2a5b1d0b-1---d8754z-;rport=54864;received=71.19.171.195
From: "101" <sip:101@domain.pbxnsip.com>;tag=a0040960
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f
Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.
CSeq: 2 INVITE
Contact: <sip:101@69.39.221.108:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: Shanticom-PBX/3.4.0.3201
Content-Type: application/sdp
Content-Length: 226

v=0
o=- 1929781437 1929781437 IN IP4 69.39.221.108
s=-
c=IN IP4 69.39.221.108
t=0 0
m=audio 57948 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=sendrecv
[7] 2011/01/26 15:45:21:	SIP Tx udp:71.19.171.195:54864:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-0926993c2a5b1d0b-1---d8754z-;rport=54864;received=71.19.171.195
From: "101" <sip:101@domain.pbxnsip.com>;tag=a0040960
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f
Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.
CSeq: 2 INVITE
Contact: <sip:101@69.39.221.108:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: Shanticom-PBX/3.4.0.3201
Content-Length: 0

[7] 2011/01/26 15:45:21:	SIP Rx udp:71.19.171.195:54864:
ACK sip:4168885555@domain.pbxnsip.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-0926993c2a5b1d0b-1---d8754z-;rport
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f
From: "101"<sip:101@domain.pbxnsip.com>;tag=a0040960
Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.
CSeq: 2 ACK
Content-Length: 0

[7] 2011/01/26 15:45:21:	Other Ports: 25
[7] 2011/01/26 15:45:21:	Call Port: 13ccde2c-b65f537a@localhost#aa6c08a0f1
[7] 2011/01/26 15:45:21:	Call Port: 154d3f0c@pbx#1008184428
[7] 2011/01/26 15:45:21:	Call Port: 3c26992c6726-eu3cfqmr3aaa#d9f059f594
[7] 2011/01/26 15:45:21:	Call Port: 3c27bb5785a3-844ug2wf59xq#b1d600805f
[7] 2011/01/26 15:45:21:	Call Port: 3c27dbc54ecc-8a0txagwtu0c#36f3df207b
[7] 2011/01/26 15:45:21:	Call Port: 3c2fcaf65979-9e1r50rqa8om#bb2b24f3c5
[7] 2011/01/26 15:45:21:	Call Port: 3c32acca4ec5-07rrezqxi0lj#af0a49a366
[7] 2011/01/26 15:45:21:	Call Port: 3c3621cf732d-2bgalyw4iq1a#cde4b05959
[7] 2011/01/26 15:45:21:	Call Port: 3c3e5313785b-2yqj8rn1fmsx#655803ba5b
[7] 2011/01/26 15:45:21:	Call Port: 3c786c9e47bb-3ihpx3ca835r#9e907d2e63
[7] 2011/01/26 15:45:21:	Call Port: 3c92cf90a4a6-u9pjbehd2t50#a0a42bee81
[7] 2011/01/26 15:45:21:	Call Port: 4169e4ea@pbx#1720057541
[7] 2011/01/26 15:45:21:	Call Port: 47678ca033aa336e69f6d1bf7e94783e@69.39.221.107#dc142f7561
[7] 2011/01/26 15:45:21:	Call Port: 5dadb756@pbx#2108676965
[7] 2011/01/26 15:45:21:	Call Port: 643386be@pbx#896059230
[7] 2011/01/26 15:45:21:	Call Port: 6e80920f6f6840490ab2826670af2f07@69.39.221.107#d5697e696f
[7] 2011/01/26 15:45:21:	Call Port: 74e7e54c38acf7c8602305190862a751@69.39.221.107#eecc166ad8
[7] 2011/01/26 15:45:21:	Call Port: 7c6e218f@pbx#887543378
[7] 2011/01/26 15:45:21:	Call Port: 8014479f@pbx#1820803417
[7] 2011/01/26 15:45:21:	Call Port: 86e63a85@pbx#1919082329
[7] 2011/01/26 15:45:21:	Call Port: 91ec0f63@pbx#280178641
[7] 2011/01/26 15:45:21:	Call Port: a0f3dd41@pbx#182301236
[7] 2011/01/26 15:45:21:	Call Port: c1f21b1eeff01a3d#241ad7d39c
[7] 2011/01/26 15:45:21:	Call Port: caf298fa@pbx#290526904
[7] 2011/01/26 15:45:21:	Call Port: ffea8091@pbx#965068692
[7] 2011/01/26 15:45:21:	SIP Rx tcp:174.89.48.178:2116:



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Well, the PBX does not treat the incoming INVITE above as a trunk call. It cannot find it in the 98 trunks and therefore it challenges it. So the snom also does not respond with another INVITE?! This would be very strange. Blacklisting was not in version 3, we can exclude that.

 

So from a PBX perspective, the PBX says: Okay, extension you want to call a number? Authenticate first, please. And that's the end of the conversation...

 

 

The call should go thru the DP which is forced onto the Asterisk <-> PBXnSIP gateway trunk...

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Oh, now I get it. The call gets challenged on the trunk side, and the PBX does not answer it.

 

Did you provide a username and a password to the trunk that is sending the INVITE out? Otherwise the PBX cannot answer the challenge. You should see something like this in the log "Answer challenge with username xxx" on log level 8 (SIP).

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Oh, now I get it. The call gets challenged on the trunk side, and the PBX does not answer it.

 

Did you provide a username and a password to the trunk that is sending the INVITE out? Otherwise the PBX cannot answer the challenge. You should see something like this in the log "Answer challenge with username xxx" on log level 8 (SIP).

 

 

 

I added an username and password, both "chris" in this case. I'm getting the same result, this is a SIP Gateway tho, I'm not used to adding credentials to it. I usually create a gateway using the IP address being trusted on both sides.

 

Where:

69.39.221.107 - asterisk pbx

69.39.221.108 - pbxnsip

domain.pbxnsip.com - domain name (valid!)

4168885555 - number trying to dial

71.19.171.195 - phone external ip

192.168.1.110 - phone internal ip

 

 


[9] 2011/01/26 16:29:08:	SIP Rx udp:71.19.171.195:54864:
INVITE sip:4168885555@domain.pbxnsip.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-2d66df632a075225-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:101@192.168.1.110:54864>
To: "4168885555"<sip:4168885555@domain.pbxnsip.com>
From: "101"<sip:101@domain.pbxnsip.com>;tag=8b16475d
Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 216

v=0
o=- 5 2 IN IP4 192.168.1.110
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.110
t=0 0
m=audio 31742 RTP/AVP 107 0 8 101
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
[9] 2011/01/26 16:29:08:	UDP: Opening socket on 0.0.0.0:57496
[9] 2011/01/26 16:29:08:	UDP: Opening socket on 0.0.0.0:57497
[9] 2011/01/26 16:29:08:	UDP: Opening socket on [::]:57496
[9] 2011/01/26 16:29:08:	UDP: Opening socket on [::]:57497
[8] 2011/01/26 16:29:08:	Could not find a trunk (98 trunks)
[8] 2011/01/26 16:29:08:	Using outbound proxy sip:71.19.171.195:54864;transport=udp because UDP packet source did not match the via header
[9] 2011/01/26 16:29:08:	Resolve 57819391: udp 71.19.171.195 54864
[9] 2011/01/26 16:29:08:	SIP Tx udp:71.19.171.195:54864:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-2d66df632a075225-1---d8754z-;rport=54864;received=71.19.171.195
From: "101" <sip:101@domain.pbxnsip.com>;tag=8b16475d
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc
Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.
CSeq: 1 INVITE
Content-Length: 0

[9] 2011/01/26 16:29:08:	Resolve 57819392: udp 71.19.171.195 54864
[9] 2011/01/26 16:29:08:	SIP Tx udp:71.19.171.195:54864:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-2d66df632a075225-1---d8754z-;rport=54864;received=71.19.171.195
From: "101" <sip:101@domain.pbxnsip.com>;tag=8b16475d
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc
Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.
CSeq: 1 INVITE
User-Agent: Shanticom-PBX/3.4.0.3201
WWW-Authenticate: Digest realm="domain.pbxnsip.com",nonce="adde8e1d5f44cfea2b9611dfaf9d361d",domain="sip:4168885555@domain.pbxnsip.com",algorithm=MD5
Content-Length: 0

[9] 2011/01/26 16:29:08:	SIP Rx udp:71.19.171.195:54864:
ACK sip:4168885555@domain.pbxnsip.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-2d66df632a075225-1---d8754z-;rport
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc
From: "101"<sip:101@domain.pbxnsip.com>;tag=8b16475d
Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.
CSeq: 1 ACK
Content-Length: 0

[9] 2011/01/26 16:29:08:	SIP Rx udp:71.19.171.195:54864:
INVITE sip:4168885555@domain.pbxnsip.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-8908ca05bb1ea447-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:101@192.168.1.110:54864>
To: "4168885555"<sip:4168885555@domain.pbxnsip.com>
From: "101"<sip:101@domain.pbxnsip.com>;tag=8b16475d
Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username="101",realm="domain.pbxnsip.com",nonce="adde8e1d5f44cfea2b9611dfaf9d361d",uri="sip:4168885555@domain.pbxnsip.com",response="e4bed3811e9b62ac982d4c98f257f5b3",algorithm=MD5
Content-Length: 216

v=0
o=- 5 2 IN IP4 192.168.1.110
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.110
t=0 0
m=audio 31742 RTP/AVP 107 0 8 101
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
[8] 2011/01/26 16:29:08:	Tagging request with existing tag
[6] 2011/01/26 16:29:08:	Sending RTP for NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.#2c0f1a95dc to 192.168.1.110:31742
[9] 2011/01/26 16:29:08:	Resolve 57819393: udp 71.19.171.195 54864
[9] 2011/01/26 16:29:08:	SIP Tx udp:71.19.171.195:54864:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-8908ca05bb1ea447-1---d8754z-;rport=54864;received=71.19.171.195
From: "101" <sip:101@domain.pbxnsip.com>;tag=8b16475d
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc
Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.
CSeq: 2 INVITE
Content-Length: 0

[8] 2011/01/26 16:29:08:	Play audio_moh/noise.wav
[9] 2011/01/26 16:29:08:	UDP: Opening socket on 0.0.0.0:50634
[9] 2011/01/26 16:29:08:	UDP: Opening socket on 0.0.0.0:50635
[9] 2011/01/26 16:29:08:	UDP: Opening socket on [::]:50634
[9] 2011/01/26 16:29:08:	UDP: Opening socket on [::]:50635
[9] 2011/01/26 16:29:08:	Resolve 57819394: url sip:69.39.221.107
[9] 2011/01/26 16:29:08:	Resolve 57819394: udp 69.39.221.107 5060
[9] 2011/01/26 16:29:08:	SIP Tx udp:69.39.221.107:5060:
INVITE sip:14168885555@69.39.221.107;user=phone SIP/2.0
Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-32227ba0f22c581d832378af1cf7a841;rport
From: <sip:chris@69.39.221.107>;tag=555707179
To: <sip:14168885555@69.39.221.107;user=phone>
Call-ID: 3fbbdc88@pbx
CSeq: 32252 INVITE
Max-Forwards: 70
Contact: <sip:chris@69.39.221.108:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: Shanticom-PBX/3.4.0.3201
Remote-Party-ID: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;party=calling;screen=yes
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 1042925789 1042925789 IN IP4 69.39.221.108
s=-
c=IN IP4 69.39.221.108
t=0 0
m=audio 50634 RTP/AVP 0 18 101
a=rtpmap:0 pcmu/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[9] 2011/01/26 16:29:08:	SIP Rx udp:69.39.221.107:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-32227ba0f22c581d832378af1cf7a841;received=69.39.221.108;rport=5060
From: <sip:chris@69.39.221.107>;tag=555707179
To: <sip:14168885555@69.39.221.107;user=phone>;tag=as69468af9
Call-ID: 3fbbdc88@pbx
CSeq: 32252 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b52bade"
Content-Length: 0

[8] 2011/01/26 16:29:08:	Answer challenge with username chris
[9] 2011/01/26 16:29:08:	Resolve 57819395: udp 69.39.221.107 5060 udp:1
[9] 2011/01/26 16:29:08:	SIP Tx udp:69.39.221.107:5060:
ACK sip:14168885555@69.39.221.107;user=phone SIP/2.0
Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-32227ba0f22c581d832378af1cf7a841;rport
From: <sip:chris@69.39.221.107>;tag=555707179
To: <sip:14168885555@69.39.221.107;user=phone>;tag=as69468af9
Call-ID: 3fbbdc88@pbx
CSeq: 32252 ACK
Max-Forwards: 70
Content-Length: 0

[9] 2011/01/26 16:29:08:	Resolve 57819396: udp 69.39.221.107 5060 udp:1
[9] 2011/01/26 16:29:08:	SIP Tx udp:69.39.221.107:5060:
INVITE sip:14168885555@69.39.221.107;user=phone SIP/2.0
Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-7525da7025674f3209d59439aaa205b0;rport
From: <sip:chris@69.39.221.107>;tag=555707179
To: <sip:14168885555@69.39.221.107;user=phone>
Call-ID: 3fbbdc88@pbx
CSeq: 32253 INVITE
Max-Forwards: 70
Contact: <sip:chris@69.39.221.108:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: Shanticom-PBX/3.4.0.3201
Remote-Party-ID: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;party=calling;screen=yes
Authorization: Digest realm="asterisk",nonce="1b52bade",response="00af5db02b5683f75aaa6b4b6abaa868",username="chris",uri="sip:14168885555@69.39.221.107;user=phone",algorithm=MD5
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 1042925789 1042925789 IN IP4 69.39.221.108
s=-
c=IN IP4 69.39.221.108
t=0 0
m=audio 50634 RTP/AVP 0 18 101
a=rtpmap:0 pcmu/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[9] 2011/01/26 16:29:08:	Message repetition, packet dropped
[6] 2011/01/26 16:29:08:	Send codec pcmu/8000
[9] 2011/01/26 16:29:08:	Resolve 57819397: udp 71.19.171.195 54864
[9] 2011/01/26 16:29:08:	SIP Tx udp:71.19.171.195:54864:
SIP/2.0 183 Ringing
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-8908ca05bb1ea447-1---d8754z-;rport=54864;received=71.19.171.195
From: "101" <sip:101@domain.pbxnsip.com>;tag=8b16475d
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc
Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.
CSeq: 2 INVITE
Contact: <sip:101@69.39.221.108:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: Shanticom-PBX/3.4.0.3201
Content-Type: application/sdp
Content-Length: 224

v=0
o=- 643243467 643243467 IN IP4 69.39.221.108
s=-
c=IN IP4 69.39.221.108
t=0 0
m=audio 57496 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=sendrecv
[9] 2011/01/26 16:29:08:	SIP Rx udp:69.39.221.107:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-7525da7025674f3209d59439aaa205b0;received=69.39.221.108;rport=5060
From: <sip:chris@69.39.221.107>;tag=555707179
To: <sip:14168885555@69.39.221.107;user=phone>;tag=as69468af9
Call-ID: 3fbbdc88@pbx
CSeq: 32253 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

[7] 2011/01/26 16:29:08:	Call 3fbbdc88@pbx#555707179: Clear last INVITE
[9] 2011/01/26 16:29:08:	Resolve 57819398: url sip:69.39.221.107
[9] 2011/01/26 16:29:08:	Resolve 57819398: udp 69.39.221.107 5060
[9] 2011/01/26 16:29:08:	SIP Tx udp:69.39.221.107:5060:
ACK sip:14168885555@69.39.221.107;user=phone SIP/2.0
Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-7525da7025674f3209d59439aaa205b0;rport
From: <sip:chris@69.39.221.107>;tag=555707179
To: <sip:14168885555@69.39.221.107;user=phone>;tag=as69468af9
Call-ID: 3fbbdc88@pbx
CSeq: 32253 ACK
Max-Forwards: 70
Contact: <sip:chris@69.39.221.108:5060;transport=udp>
Remote-Party-ID: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;party=calling;screen=yes
Content-Length: 0

[5] 2011/01/26 16:29:08:	INVITE Response 403 Forbidden: Terminate 3fbbdc88@pbx
[7] 2011/01/26 16:29:08:	Other Ports: 19
[7] 2011/01/26 16:29:08:	Call Port: 0406c8b0115619be07d4a54162592a73@69.39.221.107#c1a15e31c5
[7] 2011/01/26 16:29:08:	Call Port: 0c30e7c20f4069af55e343c81824ca9e@69.39.221.107#8da065db10
[7] 2011/01/26 16:29:08:	Call Port: 1b3599ef66a7e4567fa425996a543e61@69.39.221.107#08a8fe570c
[7] 2011/01/26 16:29:08:	Call Port: 306de8f13f896b5f720367a96baa0414@69.39.221.107#7e8c0f8c53
[7] 2011/01/26 16:29:08:	Call Port: 385a31350d24d8562e3d82110bf04fd4@69.39.221.107#2e1b3e63a7
[7] 2011/01/26 16:29:08:	Call Port: 3c269413038d-boh36yl7bz54#03bcbecf66
[7] 2011/01/26 16:29:08:	Call Port: 3c26e15be1ad-zczh8x7jb536#e59c99473a
[7] 2011/01/26 16:29:08:	Call Port: 3c2927e27a18-hm45ek6vgh3z#83f759db21
[7] 2011/01/26 16:29:08:	Call Port: 3c32b54b9ba2-rdxxbojm84pm#12cb5af3d7
[7] 2011/01/26 16:29:08:	Call Port: 4de90dff1531fcac2a40a4743cb85d7d@69.39.221.107#2cbb247a6d
[7] 2011/01/26 16:29:08:	Call Port: 72579d5c6d76566371675a317073ef73@69.39.221.107#60863285c4
[7] 2011/01/26 16:29:08:	Call Port: 74a86d7b@pbx#2030213476
[7] 2011/01/26 16:29:08:	Call Port: 97656269@pbx#1234486799
[7] 2011/01/26 16:29:08:	Call Port: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.#2c0f1a95dc
[7] 2011/01/26 16:29:08:	Call Port: a294e6ec@pbx#551257555
[7] 2011/01/26 16:29:08:	Call Port: ad6bb336@pbx#1558453222
[7] 2011/01/26 16:29:08:	Call Port: b274dc3f@pbx#1490165311
[7] 2011/01/26 16:29:08:	Call Port: bb903a2c9d68a855#8614b5139f
[7] 2011/01/26 16:29:08:	Call Port: f02665db@pbx#907681560
[9] 2011/01/26 16:29:08:	Resolve 57819399: udp 71.19.171.195 54864
[9] 2011/01/26 16:29:08:	SIP Tx udp:71.19.171.195:54864:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-8908ca05bb1ea447-1---d8754z-;rport=54864;received=71.19.171.195
From: "101" <sip:101@domain.pbxnsip.com>;tag=8b16475d
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc
Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.
CSeq: 2 INVITE
Contact: <sip:101@69.39.221.108:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: Shanticom-PBX/3.4.0.3201
Content-Length: 0

[9] 2011/01/26 16:29:09:	SIP Rx udp:71.19.171.195:54864:
ACK sip:4168885555@domain.pbxnsip.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-8908ca05bb1ea447-1---d8754z-;rport
To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc
From: "101"<sip:101@domain.pbxnsip.com>;tag=8b16475d
Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.
CSeq: 2 ACK
Content-Length: 0

[7] 2011/01/26 16:29:09:	Other Ports: 18
[7] 2011/01/26 16:29:09:	Call Port: 0406c8b0115619be07d4a54162592a73@69.39.221.107#c1a15e31c5
[7] 2011/01/26 16:29:09:	Call Port: 0c30e7c20f4069af55e343c81824ca9e@69.39.221.107#8da065db10
[7] 2011/01/26 16:29:09:	Call Port: 1b3599ef66a7e4567fa425996a543e61@69.39.221.107#08a8fe570c
[7] 2011/01/26 16:29:09:	Call Port: 306de8f13f896b5f720367a96baa0414@69.39.221.107#7e8c0f8c53
[7] 2011/01/26 16:29:09:	Call Port: 385a31350d24d8562e3d82110bf04fd4@69.39.221.107#2e1b3e63a7
[7] 2011/01/26 16:29:09:	Call Port: 3c269413038d-boh36yl7bz54#03bcbecf66
[7] 2011/01/26 16:29:09:	Call Port: 3c26e15be1ad-zczh8x7jb536#e59c99473a
[7] 2011/01/26 16:29:09:	Call Port: 3c2927e27a18-hm45ek6vgh3z#83f759db21
[7] 2011/01/26 16:29:09:	Call Port: 3c32b54b9ba2-rdxxbojm84pm#12cb5af3d7
[7] 2011/01/26 16:29:09:	Call Port: 4de90dff1531fcac2a40a4743cb85d7d@69.39.221.107#2cbb247a6d
[7] 2011/01/26 16:29:09:	Call Port: 72579d5c6d76566371675a317073ef73@69.39.221.107#60863285c4
[7] 2011/01/26 16:29:09:	Call Port: 74a86d7b@pbx#2030213476
[7] 2011/01/26 16:29:09:	Call Port: 97656269@pbx#1234486799
[7] 2011/01/26 16:29:09:	Call Port: a294e6ec@pbx#551257555
[7] 2011/01/26 16:29:09:	Call Port: ad6bb336@pbx#1558453222
[7] 2011/01/26 16:29:09:	Call Port: b274dc3f@pbx#1490165311
[7] 2011/01/26 16:29:09:	Call Port: bb903a2c9d68a855#8614b5139f
[7] 2011/01/26 16:29:09:	Call Port: f02665db@pbx#907681560
[9] 2011/01/26 16:29:09:	SIP Rx udp:174.119.245.84:60929:

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[9] 2011/01/26 16:29:08: SIP Rx udp:69.39.221.107:5060:

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-7525da7025674f3209d59439aaa205b0;received=69.39.221.108;rport=5060

From: <sip:chris@69.39.221.107>;tag=555707179

To: <sip:14168885555@69.39.221.107;user=phone>;tag=as69468af9

Call-ID: 3fbbdc88@pbx

CSeq: 32253 INVITE

User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Content-Length: 0

 

 

Seems like the Asterisk now accepts the call; however there must be something on the Asterisk that denies the call. From the PBX perspective, it looks "beautiful" now... Yes also gateway trunks can answer challenges!

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Seems like the Asterisk now accepts the call; however there must be something on the Asterisk that denies the call. From the PBX perspective, it looks "beautiful" now... Yes also gateway trunks can answer challenges!

 

Yes, seems to work fine on the PBX side. Have you ever encountered anyone using Asterisk to create a SIP Gateway with a PBXnSIP? If so, would you please post the trunk contexts?

 

I've been trying to get this to work for about a month now...

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