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Summary:ASTERISK-04063: oSIP clients cannot call through Asterisk that uses non-default SIP port
Reporter:Jonne Kodu (jkodu)Labels:
Date Opened:2005-05-04 09:00:30Date Closed:2005-05-09 08:57:49
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:My sip.conf:
[general]
port=5070

I resigter my Asterisk towards an PSTN Gateway- to witch I route all external numbers. This works perfectly when Asterisk uses SIP port 5060.

Since I, for some setups, have to use another SIP port, I configure it to 5070.
With my client, based on oSIP, it does NOT work to call external PSTN numbers. (It works to call local users.)

In additional info I attach my Asterisk output, with LinPhone making a PSTN call that fails.

****** ADDITIONAL INFORMATION ******

******************************************************************
linphone -Asterisk - PSTN Gateway (verbosity=4) Loop detected
******************************************************************


Sip read:
INVITE sip:90510@192.168.0.10:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.66:5060;branch=z9hG4bK4146386179
From: <sip:wbx@192.168.0.10>;tag=139007210
To: <sip:90510@192.168.0.10:5070>
Call-ID: 1449878670@192.168.0.66
CSeq: 20 INVITE
Contact: <sip:wbx@192.168.0.66>
max-forwards: 10
user-agent: oSIP/Linphone-0.10.2
Content-Type: application/sdp
Content-Length:   249

v=0
o=wbx 123456 654321 IN IP4 192.168.0.66
s=A conversation
c=IN IP4 192.168.0.66
t=0 0
m=audio 7078 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

11 headers, 11 lines
Using latest request as basis request
Sending to 192.168.0.66 : 5060 (non-NAT)
May  4 15:28:22 DEBUG[22546]: chan_sip.c:5437 check_user_full: Setting NAT on RTP to 4
Found user 'wbx'
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.66:7078
May  4 15:28:22 DEBUG[22546]: chan_sip.c:2711 process_sdp: Peer audio RTP is at port 192.168.0.66:7078
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
May  4 15:28:22 DEBUG[22546]: chan_sip.c:7325 handle_request: Check for res for wbx
May  4 15:28:22 DEBUG[22546]: chan_sip.c:1620 update_user_counter: Call from user 'wbx' is 1 out of 0
Looking for 90510 in incoming-sip
May  4 15:28:22 DEBUG[22546]: chan_sip.c:4584 build_route:
May  4 15:28:22 DEBUG[22546]: chan_sip.c:4646 build_route: build_route: Contact hop: <sip:wbx@192.168.0.66>
list_route: hop: <sip:wbx@192.168.0.66>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.66:5060;branch=z9hG4bK4146386179;received=192.168.0.66;rport=5060
From: <sip:wbx@192.168.0.10>;tag=139007210
To: <sip:90510@192.168.0.10:5070>;tag=as304b48a0
Call-ID: 1449878670@192.168.0.66
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:90510@192.168.0.10:5070>
Content-Length: 0


to 192.168.0.66:5060
   -- Executing SetGlobalVar("SIP/wbx-4f48", "sipto=90510") in new stack
   -- Setting global variable 'sipto' to '90510'
   -- Executing SetGlobalVar("SIP/wbx-4f48", "sipdom=192.168.0.10:5070") in new stack
   -- Setting global variable 'sipdom' to '192.168.0.10:5070'
May  4 15:28:22 DEBUG[22656]: pbx.c:1192 pbx_substitute_variables_helper: Expression is '0'
   -- Executing GotoIf("SIP/wbx-4f48", "0?30|1:5|1") in new stack
   -- Goto (incoming-sip,5,1)
May  4 15:28:22 DEBUG[22656]: pbx.c:1192 pbx_substitute_variables_helper: Expression is '0'
May  4 15:28:22 DEBUG[22656]: pbx.c:1192 pbx_substitute_variables_helper: Expression is '0'
May  4 15:28:22 DEBUG[22656]: pbx.c:1192 pbx_substitute_variables_helper: Expression is '0'
May  4 15:28:22 DEBUG[22656]: pbx.c:1192 pbx_substitute_variables_helper: Expression is '0'
   -- Executing GotoIf("SIP/wbx-4f48", "0?20|1:10|1") in new stack
   -- Goto (incoming-sip,10,1)
   -- Executing Dial("SIP/wbx-4f48", "SIP/90510@192.168.0.10:5070") in new stack
May  4 15:28:22 DEBUG[22656]: chan_sip.c:1487 sip_call: Outgoing Call for 90510
May  4 15:28:22 DEBUG[22656]: chan_sip.c:1592 update_user_counter: 90510 is not a local user
We're at 192.168.0.10 port 17770
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:90510@192.168.0.10:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5070;branch=z9hG4bK2d822d6a
From: "wbx" <sip:2010@192.168.0.10:5070>;tag=as5506f3ef
To: <sip:90510@192.168.0.10:5070>
Contact: <sip:2010@192.168.0.10:5070>
Call-ID: 7c61af0709c53a37401929c050303f38@192.168.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 04 May 2005 13:28:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 22656 22656 IN IP4 192.168.0.10
s=session
c=IN IP4 192.168.0.10
t=0 0
m=audio 17770 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 - - - -
(no NAT) to 192.168.0.10:5070


Sip read:
INVITE sip:90510@192.168.0.10:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5070;branch=z9hG4bK2d822d6a
From: "wbx" <sip:2010@192.168.0.10:5070>;tag=as5506f3ef
To: <sip:90510@192.168.0.10:5070>
Contact: <sip:2010@192.168.0.10:5070>
Call-ID: 7c61af0709c53a37401929c050303f38@192.168.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 04 May 2005 13:28:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 22656 22656 IN IP4 192.168.0.10
s=session
c=IN IP4 192.168.0.10
t=0 0
m=audio 17770 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 - - - -

12 headers, 11 lines
May  4 15:28:22 DEBUG[22546]: chan_sip.c:7794 sipsock_read: Failed to grab lock, trying again...
   -- Called 90510@192.168.0.10:5070
Transmitting (no NAT):
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 192.168.0.10:5070;branch=z9hG4bK2d822d6a
From: "wbx" <sip:2010@192.168.0.10:5070>;tag=as5506f3ef
To: <sip:90510@192.168.0.10:5070>;tag=as5506f3ef
Call-ID: 7c61af0709c53a37401929c050303f38@192.168.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2010@192.168.0.10:5070>
Content-Length: 0


to 192.168.0.10:5070


Sip read:
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 192.168.0.10:5070;branch=z9hG4bK2d822d6a
From: "wbx" <sip:2010@192.168.0.10:5070>;tag=as5506f3ef
To: <sip:90510@192.168.0.10:5070>;tag=as5506f3ef
Call-ID: 7c61af0709c53a37401929c050303f38@192.168.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2010@192.168.0.10:5070>
Content-Length: 0


10 headers, 0 lines
May  4 15:28:22 DEBUG[22546]: chan_sip.c:822 __sip_ack: Acked pending invite 102
May  4 15:28:22 DEBUG[22546]: chan_sip.c:840 __sip_ack: Stopping retransmission on '7c61af0709c53a37401929c050303f38@192.168.0.10' of Request 102: Found
   -- Got SIP response 482 "Loop Detected" back from 192.168.0.10
May  4 15:28:22 DEBUG[22546]: chan_sip.c:6925 handle_response: Hairpin detected, setting up call forward for what it's worth
Transmitting:
ACK sip:90510@192.168.0.10:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5070;branch=z9hG4bK2d822d6a
From: "wbx" <sip:2010@192.168.0.10:5070>;tag=as5506f3ef
To: <sip:90510@192.168.0.10:5070>;tag=as5506f3ef
Contact: <sip:2010@192.168.0.10:5070>
Call-ID: 7c61af0709c53a37401929c050303f38@192.168.0.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 192.168.0.10:5070


Sip read:
ACK sip:90510@192.168.0.10:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5070;branch=z9hG4bK2d822d6a
From: "wbx" <sip:2010@192.168.0.10:5070>;tag=as5506f3ef
To: <sip:90510@192.168.0.10:5070>;tag=as5506f3ef
Contact: <sip:2010@192.168.0.10:5070>
Call-ID: 7c61af0709c53a37401929c050303f38@192.168.0.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


9 headers, 0 lines
   -- Now forwarding SIP/wbx-4f48 to 'Local/90510@sip' (thanks to SIP/192.168.0.10:5070-baff)
May  4 15:28:22 NOTICE[22656]: chan_local.c:378 local_alloc: No such extension/context 90510@sip creating local channel
May  4 15:28:22 NOTICE[22656]: app_dial.c:232 wait_for_answer: Unable to create local channel for call forward to 'Local/90510@sip'
May  4 15:28:22 DEBUG[22656]: chan_sip.c:1716 sip_hangup: update_user_counter(90510) - decrement outUse counter
May  4 15:28:22 DEBUG[22656]: chan_sip.c:1592 update_user_counter: 90510 is not a local user
 == Everyone is busy/congested at this time
May  4 15:28:22 DEBUG[22656]: app_dial.c:1036 dial_exec: Exiting with DIALSTATUS=CHANUNAVAIL.
   -- Executing Hangup("SIP/wbx-4f48", "") in new stack
 == Spawn extension (incoming-sip, 10, 2) exited non-zero on 'SIP/wbx-4f48'
   -- Executing SetGlobalVar("SIP/wbx-4f48", "sipto=h") in new stack
   -- Setting global variable 'sipto' to 'h'
   -- Executing SetGlobalVar("SIP/wbx-4f48", "sipdom=192.168.0.10:5070") in new stack
   -- Setting global variable 'sipdom' to '192.168.0.10:5070'
May  4 15:28:22 DEBUG[22656]: pbx.c:1192 pbx_substitute_variables_helper: Expression is '1'
   -- Executing GotoIf("SIP/wbx-4f48", "1?30|1:5|1") in new stack
   -- Goto (incoming-sip,30,1)
   -- Executing Hangup("SIP/wbx-4f48", "") in new stack
 == Spawn extension (incoming-sip, 30, 1) exited non-zero on 'SIP/wbx-4f48'
May  4 15:28:22 DEBUG[22656]: chan_sip.c:1719 sip_hangup: update_user_counter(wbx) - decrement inUse counter
Reliably Transmitting (NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.66:5060;branch=z9hG4bK4146386179;received=192.168.0.66;rport=5060
From: <sip:wbx@192.168.0.10>;tag=139007210
To: <sip:90510@192.168.0.10:5070>;tag=as304b48a0
Call-ID: 1449878670@192.168.0.66
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:90510@192.168.0.10:5070>
Content-Length: 0


to 192.168.0.66:5060
wbx1*CLI>

Sip read:
ACK sip:90510@192.168.0.10:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.66:5060;branch=z9hG4bK4146386179
From: <sip:wbx@192.168.0.10>;tag=139007210
To: <sip:90510@192.168.0.10:5070>;tag=as304b48a0
Call-ID: 1449878670@192.168.0.66
CSeq: 20 ACK
Content-Length: 0
Comments:By: Kevin P. Fleming (kpfleming) 2005-05-04 12:24:13

I don't understand; you are saying that when you use port 5060, this doesn't happen? I don't see anything in this trace that has anything to do with calling to an oSIP peer, so the bug description seems incomplete or inaccurate.

Can you describe the situation (configuration, peers, etc.) more completely?
By: Brian West (bkw918) 2005-05-04 15:21:37

I smell a reinvite with the wrong port maybe?

/b
By: Jonne Kodu (jkodu) 2005-05-06 04:21:18

Trying to clarify the scenario:

1. the trace shows an INVITE from (not to) the oSIP peer, and my Asterisk(with port:5070) is supposed to send the INVITE to the SIP/PSTN Gateway, but its sends the INVITE to itself, and a Loop appears.
2. it isn't a re-INVITE issue, because the actual user configuration looks like this:
   [wbx]
   type=friend
   callerid=wbx <2010>
   nat=yes
   canreinvite=no ; !!!!!!!!!
   host=dynamic
   context=incoming-sip

Comparing with a working peer
-----------------------------
When I use e.g. SJphone as the calling peer instead the scenario works correctly (see trace below).

The obvious difference when comparing the INVITE from SJphone with the INVITE from linphone(and other oSIP peers), is that SJphone never mentions port 5070 in the SIP message, it just sends the messege to 5070.
Then Asterisk understands and sends the INVITE to the SIP/PSTN Gateway (service provider).


*********************************************************
This is a working scenario
SJphone -> Asterisk(using port:5070) -> SIP/PSTN Gateway
********************************************************

Sip read:
INVITE sip:90510@192.168.0.10 SIP/2.0
Content-Length: 337
Contact: <sip:wbx@192.168.0.53:5060>
Call-ID: E723888E-D638-40DE-A827-FDAE121FAB1B@192.168.0.53
Content-Type: application/sdp
From: "wbx"<sip:wbx@192.168.0.10>;tag=2535195421664
CSeq: 1 INVITE
Max-Forwards: 70
To: <sip:90510@192.168.0.10>
Via: SIP/2.0/UDP 192.168.0.53;rport;branch=z9hG4bKc0a800350131c9b14267ba0e00001d4800000198
User-Agent: SJphone/1.50.271d (SJ Labs)

v=0
o=- 3323082894 3323082894 IN IP4 192.168.0.53
s=SJphone
c=IN IP4 192.168.0.53
t=0 0
a=direction:active
m=audio 49172 RTP/AVP 3 97 98 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

11 headers, 15 lines

Using latest request as basis request
Sending to 192.168.0.53 : 5060 (NAT)
Apr 21 16:34:51 DEBUG[2476]: chan_sip.c:5437 check_user_full: Setting NAT on RTP to 4
Found user 'wbx'
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.53:49172
Apr 21 16:34:51 DEBUG[2476]: chan_sip.c:2711 process_sdp: Peer audio RTP is at port 192.168.0.53:49172
Found description format GSM
Found description format iLBC
Found description format iLBC
Found description format PCMA
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Apr 21 16:34:51 DEBUG[2476]: chan_sip.c:7325 handle_request: Check for res for wbx
Apr 21 16:34:51 DEBUG[2476]: chan_sip.c:1620 update_user_counter: Call from user 'wbx' is 1 out of 0
Looking for 90510 in incoming-sip
Apr 21 16:34:51 DEBUG[2476]: chan_sip.c:4584 build_route:

Apr 21 16:34:51 DEBUG[2476]: chan_sip.c:4646 build_route: build_route: Contact hop: <sip:wbx@192.168.0.53:5060>
list_route: hop: <sip:wbx@192.168.0.53:5060>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.53;branch=z9hG4bKc0a80035000000624267ba0e000063c50000019a;received=192.168.0.53;rport=5060
From: "wbx"<sip:wbx@192.168.0.10>;tag=2535195421664
To: <sip:90510@192.168.0.10>;tag=as31268dc9
Call-ID: E723888E-D638-40DE-A827-FDAE121FAB1B@192.168.0.53
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:90510@192.168.0.10:5070>
Content-Length: 0


to 192.168.0.53:5060
Apr 21 16:34:51 DEBUG[2640]: pbx.c:1192 pbx_substitute_variables_helper: Expression is '0'
Apr 21 16:34:51 DEBUG[2640]: pbx.c:1192 pbx_substitute_variables_helper: Expression is '0'
Apr 21 16:34:51 DEBUG[2640]: pbx.c:1192 pbx_substitute_variables_helper: Expression is '1'
Apr 21 16:34:51 DEBUG[2640]: pbx.c:1192 pbx_substitute_variables_helper: Expression is '0'
Apr 21 16:34:51 DEBUG[2640]: pbx.c:1192 pbx_substitute_variables_helper: Expression is '1'
Apr 21 16:34:51 DEBUG[2640]: app_dial.c:494 dial_exec: SIMPLE DIAL (NO URL)
Apr 21 16:34:51 DEBUG[2640]: chan_sip.c:1309 create_addr: Setting NAT on RTP to 0
Apr 21 16:34:51 DEBUG[2640]: chan_sip.c:1487 sip_call: Outgoing Call for 90510
Apr 21 16:34:51 DEBUG[2640]: chan_sip.c:1592 update_user_counter: 90510 is not a local user
We're at 192.168.0.10 port 10734
Answering/Requesting with root capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:90510@ipkund1.rixtelecom.se SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5070;branch=z9hG4bK3a80307e
From: "wbx" <sip:0850004086@192.168.0.10:5070>;tag=as01eecdb6
To: <sip:90510@ipkund1.rixtelecom.se>
Contact: <sip:0850004086@192.168.0.10:5070>
Call-ID: 2ecbe37c2b13c3c86970066531ad9677@192.168.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 21 Apr 2005 14:34:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 2640 2640 IN IP4 192.168.0.10
s=session
c=IN IP4 192.168.0.10
t=0 0
m=audio 10734 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 83.140.41.62:5060
Transmitting (NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.53;branch=z9hG4bKc0a80035000000624267ba0e000063c50000019a;received=192.168.0.53;rport=5060
From: "wbx"<sip:wbx@192.168.0.10>;tag=2535195421664
To: <sip:90510@192.168.0.10>;tag=as31268dc9
Call-ID: E723888E-D638-40DE-A827-FDAE121FAB1B@192.168.0.53
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:90510@192.168.0.10:5070>
Content-Length: 0


to 192.168.0.53:5060

Sip read:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.10:5070;branch=z9hG4bK10c7808a
From: "wbx" <sip:0850004086@192.168.0.10:5070>;tag=as01eecdb6
To: <sip:90510@ipkund1.rixtelecom.se>;tag=as5d186cd7
Call-ID: 2ecbe37c2b13c3c86970066531ad9677@192.168.0.10
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:90510@83.140.41.62>
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 10062 10062 IN IP4 83.140.41.62
s=session
c=IN IP4 83.140.41.62
t=0 0
m=audio 38930 RTP/AVP 8 0 97 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

11 headers, 13 lines
Apr 21 16:34:52 DEBUG[2476]: chan_sip.c:872 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2ecbe37c2b13c3c86970066531ad9677@192.168.0.10' Request 103: Found
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 83.140.41.62:38930
Apr 21 16:34:52 DEBUG[2476]: chan_sip.c:2711 process_sdp: Peer audio RTP is at port 83.140.41.62:38930
Found description format PCMA
Found description format PCMU
Found description format iLBC
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)

Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.10:5070;branch=z9hG4bK10c7808a
From: "wbx" <sip:0850004086@192.168.0.10:5070>;tag=as01eecdb6
To: <sip:90510@ipkund1.rixtelecom.se>;tag=as5d186cd7
Call-ID: 2ecbe37c2b13c3c86970066531ad9677@192.168.0.10
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:90510@83.140.41.62>
Content-Length: 0

etc.
**************************************************************
By: Brian West (bkw918) 2005-05-06 08:31:36

I see some significant differences in the two invites... the question is.. who is wrong...  Can you point to the exact part of the RFC we are not following?

/b
By: Olle Johansson (oej) 2005-05-08 03:42:39

Hmmm. Wrong conclusion on first attempt. Need to see your dial plan for this call, as well as sip.conf entries for this service provider.
By: Olle Johansson (oej) 2005-05-08 03:48:17

Looking at your debug output, we are processing the dial plan very differently and dial out in different ways. You need to debug your dial plan.
In the first case we have:
-- Executing Dial("SIP/wbx-4f48", "SIP/90510@192.168.0.10:5070") in new stack
So the 5070 comes from your dial plan.

In the second case we have:
Apr 21 16:34:51 DEBUG[2640]: app_dial.c:494 dial_exec: SIMPLE DIAL (NO URL)
Which is different. So the question is, why do your dial plan result in a call to ourselves?
By: Jonne Kodu (jkodu) 2005-05-09 06:35:35

Here comes my asterisk configuration:

; my sip.conf:
; ************

[general]
port=5070
bindaddr=0.0.0.0
srvlookup=yes
context=sip
disallow=all            
allow=ulaw
allow=alaw
defaultexpirey=120
maxexpirey=3600

register => 0850004086:*****@ipkund1.rixtelecom.se/rix
register => 0850004526:*****@ipkund1.rixtelecom.se/rix

[rix1]
type=peer
username=0850004086
secret=*****
fromuser=0850004086
host=ipkund1.rixtelecom.se
insecure=very

[rix2]
type=peer
username=0850004526
secret=*****
fromuser=0850004526
host=ipkund1.rixtelecom.se
insecure=very

[jonne]
type=friend
callerid=jonne <1003>
canreinvite=no
host=dynamic
secret=*****
context=incoming-sip

[wbx]
type=friend
callerid=wbx <2010>
nat=yes
canreinvite=no
host=dynamic
context=incoming-sip

; my extensions.conf
; ******************

[general]
static=yes
writeprotect=no

[globals]
MYCURRENTIPADDR=192.168.0.10
MYCURRENTDOMAIN=

[sip]
exten=jonne,1,Dial(SIP/jonne)
exten=wbx,1,Dial(SIP/wbx)
exten=1003,1,Dial(Local/jonne@sip)
exten=2010,1,Dial(Local/wbx@sip)
exten=rix,1,Dial(Local/jonne@sip)

[incoming-sip]
exten=_.,1,SetGlobalVar(sipto=${EXTEN})
exten=_.,2,SetGlobalVar(sipdom=${SIPDOMAIN})
exten=_.,3,GotoIf($[${sipto} = h]?30,1:5,1)
exten=5,1,GotoIf($[$[${SIPDOMAIN} = ${MYCURRENTDOMAIN}] | $[${SIPDOMAIN} = ${MYCURRENTIPADDR}] |
$[foo${SIPDOMAIN} = foo]]?20,1:10,1)
exten=10,1,Dial(SIP/${sipto}@${sipdom})
exten=10,2,Hangup
exten=20,1,Goto(default,${sipto},1)
exten=30,1,Hangup

[outgoing-sip]
exten=_X.,1,Dial(SIP/${EXTEN}@rix1,,r)
exten=_X.,2,Hangup

[incoming-pri]
include=sip

[default]
include=sip
include=outgoing-sip
By: Jonne Kodu (jkodu) 2005-05-09 07:04:10

Thanks Olle for the dialplan hint! I found the problem. :-)

If I add the 5070 port to the MYCURRENTIPADDR global, then it works with oSIP peers:

[globals]
MYCURRENTIPADDR=192.168.0.10:5070