[asterisk-users] problems with some incoming/outgoing calls

[Fernando Serto]

Hi, Sorry I forgot to mention versions and post files.

Asterisk version:
pbx:/etc/asterisk# asterisk -rx "core show version"
Asterisk 1.4.22 built by root @ coope-pbx on a i686 running Linux on
2008-10-22 09:36:35 UTC

I'm running zaptel 1.4.12.1 and wanpipe 3.3.14. Also tried zaptel
1.4.11 and 1.4.12, and wanpipe 3.2.7.1, and the problem happens on all
versions.

extensions.conf: http://www.pastebin.ca/1235312
sip.conf: http://www.pastebin.ca/1235317
zapata.conf: http://www.pastebin.ca/1235318
zaptel.conf: http://www.pastebin.ca/1235322

Let me know if I should be posting any other conf files.

On Fri, Oct 24, 2008 at 1:27 AM, Lucas Alvarez <lucas at provetel.com.ar> wrote:
> Hi, which version of asterisk are you running? Perhaps if you post your
> extensions.conf and others related files you could get more accurate help.
> If you answer a ringing phone and you can't answer the call, there you
> could have a network or sip config problem, that means that the SIP packet
> is not returning to the pbx.
> Regards.
>
> Lucas
>
>
> On Thu, 23 Oct 2008 10:30:29 -0200, Fernando Serto <fserto at gmail.com>
> wrote:
>
>> Hi,
>>
>> I've been very puzzled lately. I installed a phone system for a friend
>> a few weeks ago, and they're having a problem that I can't get rid of,
>> actually 2 problems. Before I go into the problems, let me tell you
>> about the setup. It's a pretty small setup with only 4 handsets, all
>> Polycom 320s, the server is a Dell SC440 with Intel E2180 CPU (dual
>> core, 2GHz) and 512MB Ram. Internet Connection is an ADSL2, with a not
>> so reliable ISP in australia. For incoming calls, I had a Digium
>> TDM410P with 4xFXO modules and HWEC. Because of these problems, i
>> replaced the Digium card with a Sangoma A200D, but it didn't make any
>> difference to the problems. All phones are hooked up to a Netgear PoE
>> switch.
>>
>> Almost forgot to mention that this is not my first Asterisk setup, and
>> in fact it is my 4th, and I used various SIP handsets before, and also
>> different cards (Analog and Digital), so I'm not a total noob.
>>
>> Let's get to the problems...
>>
>> 1) Some incoming calls cannot be picked up
>> Sometimes, incoming calls, coming through the analog card, cannot be
>> picked up. All handsets are set to ring at the same time on incoming
>> calls. and most of the time, calls can be answered on any of the
>> handsets, but maybe 3 or 4 times a day, all handsets will be ringing,
>> and you go to one handset to answer the call, you pick the handset,
>> and it doesn't answer the call, it keeps ringing, then you go to
>> another handset, and still can't pick up, sometimes, you can even try
>> all 4 handsets, and no luck. but, at other times, you can't answer on
>> the first handset, but you can on another, and it is totally random.
>> but people are pretty pissed off for running around to answer a call.
>> and what puzzles me is that you can sit around watching logs for
>> hours, and it won't happen, other times, it happens 3 times in a row.
>> any ideas?
>>
>> 2) Delay on outgoing calls via SIP
>> People have been saying that when they call people, there's a delay
>> for the call to be answered. For example, caller dials a number,
>> callee answers the ringing phone, but caller is still listening to a
>> ringing tone, and after a few seconds (up to 15 seconds) it sounds
>> like the callee has just answered the call, when in fact, he had
>> already answered a few seconds before. Problem with this is that some
>> callees will hangup before the caller starts talking. These calls are
>> going via pennytel, in australia, which seems to be a pretty good VOIP
>> provider around here, and I've been using it on other setups and never
>> had these issues.
>>
>> Well, sorry for the long first post, but I would really appreciate any
>> suggestions you have.
>>
>> Cheers,
>> Fernando
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>