[Asterisk-Users] Calls dropping off
Andres
andres at telesip.net
Fri Feb 6 18:18:21 MST 2004
Steve Foy wrote:
>Right... It just happened there now, this came up:
>
>Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call 73f2d12d3ab7dc271e41f3e51990ff11 at 212.3.160.87 for seqno 3 (Response)
>
>
>
So did it drop a few seconds into the call...like 5 - 15 seconds? If so
then you are having a problem with call setup. I would guess it is the
ACK that is not receiving a STATUS 200 OK so Asterisk cuts off the call.
>I'm not sure if that's related to it, but it's the only thing that came up
>when the call got cut off.
>
>Here's the generic sip.conf stuff
>
>[general]
>port = 5060 ; Port to bind to (SIP is 5060)
>bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
>
>allow=all
>allow=GSM
>allow=G729
>allow=iLBC
>allow=SpeeX ; Allow all codecs
>allow=ulaw
>
>Here's a sip.conf declaration:
>
>; Andy
>[108]
>type=friend
>username=********
>secret=********
>host=dynamic
>dtmfmode=rfc2833
>callerid="Andy McAlister" <108>
>context=internal
>mailbox=108 at default
>qualify=yes
>canreinvite=no
>
>And the relevant extension.conf bit:
>
>;Andy
>exten => 108,1,Dial(SIP/108,15)
>exten => 108,2,Playback(int-voicemail/108)
>exten => 108,3,Voicemail(s108)
>exten => 108,102,Playback(int-voicemail/108)
>exten => 108,103,Voicemail(s108)
>
>Any insight vastly appreciated!
>
>Cheers,
>Steve
>
>
>On Thu, Feb 05, 2004 at 06:33:06AM -0600, Rich Adamson wrote:
>
>
>>Steve,
>>Since I have a rather short memory and receive about 250 posting per day, I
>>don't have a clue what has/hasn't been suggested. Here's a couple:
>>1. in logger.conf turn on debug, watch /var/log/asterisk/debug for size, and
>> and hints relative to the dropped calls
>>2. look at /var/log/asterisk/messages for hints
>>3. if the problem occurs frequently enough, start a ping from the * box to
>> one or more of the sip phones to verify you're not loosing net connections
>> at the time of the dropped call (Spanning Tree Protocol can mess with your
>> infrastructure without you knowing it, as one example)
>>4. look in /var/log/asterisk/cdr-csv/Master.csv file to see if any hints in
>> the cdr data
>>5. post a relavent definition from sip.conf so we have a clue how you've
>> defined a phone, as well as a relative Dial section from extensions.conf
>> and zapata.conf
>>6. I don't recall which sip phones you're using, but some have internal
>> logging capabilities. If your's do, turn it on and look for hints.
>>7. Download ethereal and sniff the asterisk nic interface, ensure you stop
>> it right after a failure. If you need help doing the protocol analysis,
>> then let me know.
>>
>>Rich
>>
>>------------------------
>>
>>
>>>I would have thought that if that was the problem, we couldn't makle or
>>>receive calls at all, or that we at least couldnt use all 3 Zap cards at the
>>>same time, but we can.
>>>
>>>The problem only happens every so often, but recently it's getting more and
>>>more frequent... management are starting to get pissed :/
>>>
>>>No more ideas?
>>>
>>>I've tried everything else people have mentioned.
>>>
>>>Cheers,
>>>Steve
>>>
>>>On Mon, Feb 02, 2004 at 01:03:01PM -0500, Bill Hamel wrote:
>>>
>>>
>>>>Hi,
>>>>
>>>>Have you checked for IRQ conflicts ?
>>>>
>>>>-b
>>>>
>>>>Quoting Steve Foy <steve at unite.net>:
>>>>
>>>>
>>>>
>>>>>Hi,
>>>>>
>>>>>On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote:
>>>>>
>>>>>
>>>>>>Steve,
>>>>>>
>>>>>>this really is a FAQ. You need add to EACH (!) sip user something like
>>>>>>
>>>>>>disallow=all
>>>>>>allow=ulaw
>>>>>>allow=alaw
>>>>>>allow=gsm
>>>>>>
>>>>>>
>>>>>I do have that in my sip.conf. I am using ulaw.
>>>>>
>>>>>Calls from the SIP phones through Asterisk and out one of my X100P cards are
>>>>>working 95% of the time and also, incoming calls through the X100P cards to
>>>>>the SIP phones are the same.
>>>>>
>>>>>The only problem is that every once in a while, without any odd circustances
>>>>>that I can see, the call just drops and the remote user is gone.
>>>>>
>>>>>The box running asterisk isn't under heavy load, so I can't see why this is
>>>>>happening.
>>>>>
>>>>>I am not using g.729 or 723, just plain old ulaw, which I have got enabled
>>>>>in
>>>>>sip.conf
>>>>>
>>>>>Cheers,
>>>>>Steve
>>>>>
>>>>>--
>>>>>Steve Foy | http://www.unite.net
>>>>>UNITE Solutions | Tel: 028 9077 7338
>>>>>_______________________________________________
>>>>>Asterisk-Users mailing list
>>>>>Asterisk-Users at lists.digium.com
>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>To UNSUBSCRIBE or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>>--
>>>>>This message has been scanned for viruses and
>>>>>dangerous content by The CCIS.net MailScanner, and is
>>>>>believed to be clean.
>>>>>
>>>>>
>>>>>--
>>>>>This message has been scanned for viruses and
>>>>>dangerous content by the Bugs.Hamel.Net MailScanner,
>>>>>and appears to be clean.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>--
>>>>
>>>>
>>>>----------------------------------------------------------------
>>>>This message was sent using IMP, the Internet Messaging Program.
>>>>
>>>>--
>>>>This message has been scanned for viruses and
>>>>dangerous content by the Bugs.Hamel.Net MailScanner,
>>>>and appears to be clean.
>>>>
>>>>_______________________________________________
>>>>Asterisk-Users mailing list
>>>>Asterisk-Users at lists.digium.com
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>>
>>>--
>>>Steve Foy | http://www.unite.net
>>>UNITE Solutions | Tel: 028 9077 7338
>>>_______________________________________________
>>>Asterisk-Users mailing list
>>>Asterisk-Users at lists.digium.com
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>---------------End of Original Message-----------------
>>
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>
>
>
--
Andres
Network Admin
http://www.telesip.net
More information about the asterisk-users
mailing list