[Asterisk-Users] Asterisk 1.2 problems

tneuwert@formos.com
Fri, 02 Dec 2005 12:22:52 -0000

Help! I've encountered some problems with Asterisk that I?m unable to solve.
We have been running Asterisk version 1.0.9 for many months using a few local
network connected Cisco 7960 phones as SIP clients.  All our phones are
currently internal so there is no NAT involved.  We were not having any problems
until last week when some strange issues started to crop up. I started
experiencing calls that I initially believed were being dropped, but discovered
that only one side of the conversation had dropped.  The other party could hear
me but I couldn't hear them. This seems to happen more often on longer calls
but is not consistent.  I am also seeing issues where incoming or local
extension calls that are hung up by the originator before being answered will
continue to ring the SIP phone. At the time the errors occur, the Asterisk
console displays a variety of "...retrans_pkt: Maximum retries exceeded on
call.." messages. I scoured the forums for an answer, found many reference
 s to these errors, tried every suggested fix that I could find, but none have
resolved these problems.  After working on the problem for several days, I
finally built a new box and installed Asterisk 1.2 on it. Using this new 1.2 box
I no longer see the "Maximum retries exceeded on call" warnings on the
console but still experience the strange behavior. Unfortunately, the errors
occur randomly so I am unable to reproduce the error on demand. I turned on SIP
debugging and set console logging to debug and captured an instance of the
problem with the hang up not being recognized.  The details are below:

I dial in from my cell phone. My Cisco phone begins to ring. I then hang up my
cell phone. Asterisk acknowledges the hang up, but the Cisco phone continues to
ring. After a minute or so, or if I pickup the phone, Asterisk display the
following message "That's odd...  Got a response on a call we don?t
know about. Cseq 102 Cmd SIP/2.0"  I've included a copy of the console
output when this occurs that shows both the SIP message and the Asterisk debug
output.

Let me know if any more information would be of use and thanks in advance!

The Cisco phone is on IP 192.168.2.203
The Asterisk switch is on IP 192.168.2.30


<-- SIP read from 192.168.2.203:50237:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.2.30:5060;branch=z9hG4bK3dd277f1;rport
From: "JOHN DOE " <sip:3602818233@192.168.2.30>;tag=as78389007
To: <sip:121@192.168.2.203:5060>;tag=001380df7eee002b0c2db83c-5ecedbb5
Call-ID: 7d75da0679062aa677baa849101d63e3@192.168.2.30
Date: Fri, 02 Dec 2005 17:04:49 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:121@192.168.2.203:5060>
Content-Length: 0


Dec  2 09:04:37 VERBOSE[3842] logger.c: --- (10 headers 0 lines)Dec  2 09:04:37
VERBOSE[3842] logger.c: --- (10 headers 0 lines)---
Dec  2 09:04:37 DEBUG[3842] chan_sip.c: That's odd...  Got a response on a
call we dont know about. Cseq 102 Cmd SIP/2.0