Author Topic: SIP Routing Issue: Call rejected error='noconn' reason='Invalid Address'  (Read 5603 times)

jeffl

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I have a config issue which has me perplexed (new Yate user). Any assistance from an

experienced Yate admin would be greatly appreciated!!

Issue:
A Cisco GW enabled for SIP (10.100.2.10), is sending a call to Yate (10.10.2.25:5060 tcp). I

want to configure Yate to Proxy between the GW and my Cisco Unified Communications Manager

over a SIP trunk (10.10.2.20) - no secure authorization. The call fails and the Yate debug

shows the error to be as follows:

<sip/20:WARN> Could not create party for 'sip:11117801001@10.10.2.20' [02607350]
<sip/20:ALL> YateSIPConnection::hangup() state=1 trans=00000000 error='failure' code=500

reason='Invalid address: sip:11117801001@10.10.2.20' [02607350]
<sip/20:ALL> YateSIPConnection::~YateSIPConnection() [02607350]
<sip/19:MILD> Call rejected error='noconn' reason='Invalid address:

sip:11117801001@10.10.2.20' [025FAF00]

Call & Config Details:
Details of the call flow and conf file settings below (btw - when Yate is removed, call

completes successfully):

SIP phone registered to Cisco GW (SIP) DN=13335551234
SIP phone calls 11117801001 > Dial-peer routes to 10.10.2.25:5060 tcp
Call fails on Yate

###################
accfile.conf entry
###################

[UCM]
enabled=yes
protocol=sip
description=10.10.2.20
outbound=10.10.2.20
ip_transport:tcp

#######################
regexroiute.conf entry
#######################

[Default]
${id}^sip/=if ^111.*$=sip/\0;sdp_forward=yes;line=UCM


Yate Debug Output:

<sip:ALL> Listener(TCP,'general') ':5060' got conn from '10.100.2.10:58606' [025E0D90]
<sip:ALL> Transport(tcp:10.10.2.25:5060-10.100.2.10:58606) created [025FAD68]
<sip:ALL> Transport(tcp:10.10.2.25:5060-10.100.2.10:58606) initialized maxpkt=4096

rtp_localip=(null) nat_address=(null) tcp_idle=120 [025FAD68]
<sip:INFO> 'tcp:10.10.2.25:5060-10.100.2.10:58606' received 1124 bytes SIP message

[025FAD68]
------
INVITE sip:11117801001@10.10.2.25:5060 SIP/2.0
Via: SIP/2.0/TCP 10.100.2.10:5060;branch=z9hG4bK134BD3
Remote-Party-ID: "13335551234"

<sip:13335551234@10.100.2.10>;party=calling;screen=no;privacy=off
From: "13335551234" <sip:13335551234@10.100.2.10>;tag=13C95DA4-23E7
To: <sip:11117801001@10.10.2.25>
Date: Mon, 27 Jan 2014 16:37:19 GMT
Call-ID: 1FBBE245-86A811E3-860AADEB-47D93ADA@10.100.2.10
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0532086893-2259161571-2248453611-1205418714
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO,

REGISTER
CSeq: 101 INVITE
Timestamp: 1390840639
Contact: <sip:13335551234@10.100.2.10:5060;transport=tcp>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 188

v=0
o=CiscoSystemsSIP-GW-UserAgent 7080 1532 IN IP4 10.100.2.10
s=SIP Call
c=IN IP4 10.100.2.10
t=0 0
m=audio 18158 RTP/AVP 0
c=IN IP4 10.100.2.10
a=rtpmap:0 PCMU/8000
a=ptime:20
------
<sip/19:ALL> YateSIPConnection::YateSIPConnection(026080B8,0262A860) [025FAF00]
<sip/19:ALL> NAT address is '(null)' [025FAF00]
<sip/19:ALL> Set media: audio=mulaw [025FAF00]
<sip:INFO> 'tcp:10.10.2.25:5060-10.100.2.10:58606' sending code 100 026321E8 [025FAD68]
------
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.100.2.10:5060;branch=z9hG4bK134BD3;received=10.100.2.10
From: "13335551234" <sip:13335551234@10.100.2.10>;tag=13C95DA4-23E7
To: <sip:11117801001@10.10.2.25>
Call-ID: 1FBBE245-86A811E3-860AADEB-47D93ADA@10.100.2.10
CSeq: 101 INVITE
Server: YATE/5.0.0
Content-Length: 0

------
<sip/19:ALL> NAT address is '(null)' [025FAF00]
<sip/20:ALL> YateSIPConnection::YateSIPConnection(025F9BB0,'11117801001') [02607350]
<sip:ALL> YateSIPEndPoint::buildParty(02631F60,'(null)',0,025FFB88)
<sip/20:WARN> Could not create party for 'sip:11117801001@10.10.2.20' [02607350]
<sip/20:ALL> YateSIPConnection::hangup() state=1 trans=00000000 error='failure' code=500

reason='Invalid address: sip:11117801001@10.10.2.20' [02607350]
<sip/20:ALL> YateSIPConnection::~YateSIPConnection() [02607350]
<sip/19:MILD> Call rejected error='noconn' reason='Invalid address:

sip:11117801001@10.10.2.20' [025FAF00]
<sip/19:ALL> YateSIPConnection::hangup() state=0 trans=0262A860 error='noconn' code=503

reason='Invalid address: sip:11117801001@10.10.2.20' [025FAF00]
<sip/19:ALL> YateSIPConnection::~YateSIPConnection() [025FAF00]
<sip:INFO> 'tcp:10.10.2.25:5060-10.100.2.10:58606' sending code 503 02631F60 [025FAD68]
------
SIP/2.0 503 Invalid address: sip:11117801001@10.10.2.20
Via: SIP/2.0/TCP 10.100.2.10:5060;branch=z9hG4bK134BD3;received=10.100.2.10
From: "13335551234" <sip:13335551234@10.100.2.10>;tag=13C95DA4-23E7
To: <sip:11117801001@10.10.2.25>
Call-ID: 1FBBE245-86A811E3-860AADEB-47D93ADA@10.100.2.10
CSeq: 101 INVITE
Server: YATE/5.0.0
Contact: <sip:11117801001@10.10.2.25:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

------
<sip:INFO> 'tcp:10.10.2.25:5060-10.100.2.10:58606' received 384 bytes SIP message [025FAD68]
------
ACK sip:11117801001@10.10.2.25:5060 SIP/2.0
Via: SIP/2.0/TCP 10.100.2.10:5060;branch=z9hG4bK134BD3
From: "13335551234" <sip:13335551234@10.100.2.10>;tag=13C95DA4-23E7
To: <sip:11117801001@10.10.2.25>
Date: Mon, 27 Jan 2014 16:37:19 GMT
Call-ID: 1FBBE245-86A811E3-860AADEB-47D93ADA@10.100.2.10
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

------

?????

Thanks!!!

cc08

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Re: SIP Routing Issue: Call rejected error='noconn' reason='Invalid Address'
« Reply #1 on: January 27, 2014, 11:52:51 AM »
I think you can do without accfile module.
and the regexroute.conf rewritten like this:
Code: [Select]
[default]
${ip_host}^10\.10\.2\.20$=sip/sip:${called}@10.10.2.30;...forward and so on
${ip_host}^10\.10\.2\.30$=sip/sip:${called}@10.10.2.20;...
but you not showed full log and configs...
so it may not work ...
I remember how it well worked on the UDP transport.


Best regards.

ps and please read the documentation carefully, there it is ...


marian

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Re: SIP Routing Issue: Call rejected error='noconn' reason='Invalid Address'
« Reply #2 on: January 28, 2014, 02:18:39 AM »
Can you post a yate log from startup?
Please start message sniffer also:

yate.conf:

[general]
msgsniff=yes

I would like to see the log when the line is trying to register.

jeffl

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Re: SIP Routing Issue: Call rejected error='noconn' reason='Invalid Address'
« Reply #3 on: January 28, 2014, 08:24:26 AM »
Thanks for the reply Marian. Being new to Yate and having installed on a Windows platform, can you advise on the location of this log? I can definitely enable the message sniffer and capture what you are asking for.

Thanks!

marian

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Re: SIP Routing Issue: Call rejected error='noconn' reason='Invalid Address'
« Reply #4 on: January 29, 2014, 01:55:49 AM »
The log is created by passing a -l command line parameter (extra -t will truncate the log file at each start).

Example:
yate -tl c:\yate.log

More at http://docs.yate.ro/wiki/Starting_on_windows