SIP call fails with a SIP error 484 address incomplete

Hello,

I have recently upgraded to Asterisk 1.8.3.3 running on FC15. My old configuration has stopped working on incoming SIP calls from my PSTN (via Sipura 300A).

The point of failure appears to be an “address incomplete”. Can somebody help me resolve this or where to look for more info?

The log file:

[code]<------------->
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c:
<— SIP read from UDP:192.168.1.12:5061 —>
INVITE sip:192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:5061;branch=z9hG4bK-76d09ed
From: shed-line sip:shed-line@192.168.1.5;tag=7a1dfabfe775e81do1
To: sip:192.168.1.5
Remote-Party-ID: shed-line sip:shed-line@192.168.1.5;screen=yes;party=calling
Call-ID: 315ac0e7-83bf9815@192.168.1.12
CSeq: 101 INVITE
Max-Forwards: 70
Contact: shed-line sip:shed-line@192.168.1.12:5061
Expires: 240
User-Agent: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 438
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 2385 2385 IN IP4 192.168.1.12
s=-
c=IN IP4 192.168.1.12
t=0 0
m=audio 16440 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: — (15 headers 20 lines) —
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Sending to 192.168.1.12:5061 (no NAT)
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Using INVITE request as basis request - 315ac0e7-83bf9815@192.168.1.12
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found peer ‘shed-line’ for ‘shed-line’ from 192.168.1.12:5061
[Jun 6 07:17:47] VERBOSE[7881] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found RTP audio format 0
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found RTP audio format 2
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found RTP audio format 4
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found RTP audio format 8
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found RTP audio format 18
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found RTP audio format 96
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found RTP audio format 97
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found RTP audio format 98
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found RTP audio format 100
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found RTP audio format 101
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format PCMU for ID 0
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format G726-32 for ID 2
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format G723 for ID 4
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format PCMA for ID 8
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format G729a for ID 18
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format G726-40 for ID 96
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format G726-24 for ID 97
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format G726-16 for ID 98
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format NSE for ID 100
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format telephone-event for ID 101
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10d (g723|ulaw|alaw|g729)
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Peer audio RTP is at port 192.168.1.12:16440
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Looking for in shed-line (domain 192.168.1.5)
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c:
<— Reliably Transmitting (no NAT) to 192.168.1.12:5061 —>
SIP/2.0 484 Address Incomplete

Via: SIP/2.0/UDP 192.168.1.12:5061;branch=z9hG4bK-76d09ed;received=192.168.1.12

From: shed-line sip:shed-line@192.168.1.5;tag=7a1dfabfe775e81do1

To: sip:192.168.1.5;tag=as712ba20c

Call-ID: 315ac0e7-83bf9815@192.168.1.12

CSeq: 101 INVITE

Server: Chezstephens Asterisk PBX

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

Supported: replaces, timer

Content-Length: 0

<------------>
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Scheduling destruction of SIP dialog ‘315ac0e7-83bf9815@192.168.1.12’ in 32000 ms (Method: INVITE)
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c:
<— SIP read from UDP:192.168.1.12:5061 —>
ACK sip:192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:5061;branch=z9hG4bK-76d09ed
From: shed-line sip:shed-line@192.168.1.5;tag=7a1dfabfe775e81do1
To: sip:192.168.1.5;tag=as712ba20c
Call-ID: 315ac0e7-83bf9815@192.168.1.12
CSeq: 101 ACK
Max-Forwards: 70
Contact: shed-line sip:shed-line@192.168.1.12:5061
User-Agent: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0

<------------->
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: — (10 headers 0 lines) —
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Really destroying SIP dialog ‘315ac0e7-83bf9815@192.168.1.12’ Method: ACK
[/code]

Part of sip.conf:

[shed-line] type=friend host=dynamic username=shed-line secret=<hidden> call-limit=2 disallow=all allow=ulaw allow=alaw allow=g723.1 allow=g729 allow=gsm context=shed-line canreinvite=yes insecure=port,invite

And part of extensions.conf:

[shed-line] exten => s,1,NoOp() exten => s,n,Dial(Local/SHED-PHONES@cs-extens&Local/DELAYED-HOUSE-PHONES@cs-extens,30,tTkK) exten => s,n,Voicemail(10,u) ; If unavailable, send to voicemail w/ unavail announce

I know that Local/SHED-PHONES@cs-extens works as I can call this locally.

Regards,

Adrian

The called number isn’t defaulting to “s”. That may be a bug in that version.

Hello,

Is there any way to specify the default incoming extension in the sip.conf channel? I scanned through the docs and didn’t see anything.

BR,

Adrian

The default is supposed to be “s”.

This might be a bug, or this might be a failure of myself to grok the documentation.

However, there is a workaround.

Log into the phone’s webserver, admin, advanced.
In the PSTN line, your dial plan will probably look something like (S0<:@192.168.1.5>), which says
send all incoming calls to the specified gateway address.

It looks like Asterisk is failing to detect the “no extension” (in which case it should substitute “s” for the extension).
However, we can force “s” as the incoming extension using the following syntax:
(S0<:s><:@192.168.1.5>). This says, on an incoming call, insert the “number” s and pass it to the specified
gateway.

Thanks again to Dave55 for his help.

Best Regards,

Adrian