[asterisk-users] Asterisk & Cisco calling Name

[John Bittner]

Anyone see an issue on asterisk 1.2 that it will not accept the invite
from a Cisco gateway. If I turn off voice service voip signaling
forward unconditional then Asterisk accepts the call but without cname.
Below is a trace.

Any help is appreciated.

Thanks

John Bittner
Simlab.net




voippbx01*CLI>
<-- SIP read from 216.86.35.24:63549: 
INVITE sip:9734333001 at 69.60.198.130:5060 SIP/2.0
Via: SIP/2.0/UDP  216.86.35.24:5060;x-route-tag="tgrp:PRI-TRUNK-GROUP1";branch=z9hG4bK111A56
From: <sip:9733901090 at 216.86.35.24>;tag=4F9EF08-163B
To: <sip:9734333001 at 69.60.198.130>
Date: Sat, 01 Dec 2007 05:23:25 GMT
Call-ID: 602E8F94-9F0411DC-8ACEEC29-3723F693 at 216.86.35.24
Supported: 100rel,timer,replaces
Min-SE:  1800
Cisco-Guid: 1613584196-2667844060-2152857615-892193345
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: "pending" <sip:9733901090 at 216.86.35.24>;party=calling;screen=yes;privacy=off
Timestamp: 1196486605
Contact: <sip:9733901090 at 216.86.35.24:5060>
Expires: 180
Allow-Events: telephone-event
MIME-Version: 1.0
Content-Type: multipart/mixed;boundary=uniqueBoundary
Content-Length: 680

--uniqueBoundary
Content-Type: application/sdp

v=0
o=CiscoSystemsSIP-GW-UserAgent 6852 2375 IN IP4 216.86.35.24
s=SIP Call
c=IN IP4 216.86.35.24
t=0 0
m=audio 18472 RTP/AVP 0 101
c=IN IP4 216.86.35.24
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--uniqueBoundary
Content-Type: application/gtd
Content-Disposition: signal;handling=optional

IAM,
PRN,isdn*,,NI***,
USI,rate,c,s,c,1
USI,lay1,ulaw
TMR,00
CPN,04,,1,9734333001
CGN,04,,1,y,4,9733901090
CPC,09
FCI,,,,,,,y,
GCI,602d57449f0411dc8052000f352dca41
UFC,GEN,5,gentf,79
UFC,GEN,5,fachd,9f8b0100
UFC,GEN,5,inpdu,02010106072a8648ce150004

--uniqueBoundary--

--- (21 headers 33 lines)---
Using INVITE request as basis request - 602E8F94-9F0411DC-8ACEEC29-3723F693 at 216.86.35.24
Sending to 216.86.35.24 : 5060 (non-NAT)
Found peer '216.86.35.24'
Transmitting (no NAT) to 216.86.35.24:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP  216.86.35.24:5060;x-route-tag="tgrp:PRI-TRUNK-GROUP1";branch=z9hG4bK111A56;received=216.86.35.24
From: <sip:9733901090 at 216.86.35.24>;tag=4F9EF08-163B
To: <sip:9734333001 at 69.60.198.130>;tag=as39c359be
Call-ID: 602E8F94-9F0411DC-8ACEEC29-3723F693 at 216.86.35.24
CSeq: 101 INVITE
User-Agent: SimlabVOIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9734333001 at 69.60.198.130>
Content-Length: 0


---
Destroying call '602E8F94-9F0411DC-8ACEEC29-3723F693 at 216.86.35.24'
voippbx01*CLI> 
<-- SIP read from 216.86.35.24:5060: 
ACK sip:9734333001 at 69.60.198.130:5060 SIP/2.0
Via: SIP/2.0/UDP  216.86.35.24:5060;x-route-tag="tgrp:PRI-TRUNK-GROUP1";branch=z9hG4bK111A56
From: <sip:9733901090 at 216.86.35.24>;tag=4F9EF08-163B
To: <sip:9734333001 at 69.60.198.130>;tag=as39c359be
Date: Sat, 01 Dec 2007 05:23:25 GMT
Call-ID: 602E8F94-9F0411DC-8ACEEC29-3723F693 at 216.86.35.24
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0