asterisk users - Dec 2005 - Re: Asterisk 1.2 problems (tneuwert@formos.com)

What version firmware are you running on your Cisco Phones? We are running
Asterisk 1.2 with the 7.4 firmware. The latest is 7.5 but there are some strange
things that happen with this firmware. If I were you I would try a different
firmware on the phones. Hope this helps.
Jeremiah


>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

We are using firmware version 6.3. Don?t we need a service agreement to get the
latest drivers? We let ours expire since we weren?t having any problems. Isn?t
it also true that once you upgrade the firmware there is no way to revert to an
earlier version? This is worrisome because we have heard of "bad
versions" and do not want to upgrade without having a back out plan.
Thanks,
Tim
> What version firmware are you running on your Cisco Phones? We are
> running Asterisk 1.2 with the 7.4 firmware. The latest is 7.5 but there
> are some strange things that happen with this firmware. If I were you I
> would try a different firmware on the phones. Hope this helps. Jeremiah
> 
> 
> 
>> 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
> 
> 
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