VoIP Mailing List Archives

[yves030 at gmx.de]

hi,

I have a really strange problem which is driving me crazy for days now.

If I register my asterisk (tried all versions from 1.6 up to 13.x) with
one sip registrar,
everything works... calls go out and call come in... no 32 seconds limit.

but as soon as I configure another sip registration on another server,
outgoing
calls drop after 32 seconds.

as far as I know, there is no firewall in between...

I tried to "work around" this by increasing the settings for "timerb"...
but I
realized that asterisk does not care at all, what I set this value to...
"sip show settings" always gives me 32000ms, and it does not make any
difference if I configure timerb in the general context or in the phone
context...

any ideas?

thanks,
yves

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[yves030 at gmx.de]

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:

[EWieling at nyigc.com]

Try setting directmedia=no in sip.conf.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Yves A.
Sent: Saturday, November 22, 2014 8:06 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:

[visser.rafael at gmail...]

Hi Yves..

This may be silly... but what is the useragent of your sip configuration?

In the case that useragent has some special characters like "(.", please remove it and tell us if there is any change!!.

Regards.
rv




2014-11-22 14:50 GMT-03:00 Eric Wieling <EWieling@nyigc.com (EWieling@nyigc.com)>:

[rwheeler at artifact-s...]

You might check your phones as well.
We had this problem early on with a softphone and it was a setting in
the phone that was set to hang up after 30 seconds of inactivity "in
case of network disruption". For some reason it was detecting "network
disruption" in every call even when the calls were proceeding normally.
Unchecking this box solved the problem.

It may not be related to your problem but if it is the cause, you will
spend a lot of time trying to fix this in Asterisk. At least I did!

On the bright side, it does force people to get point in a hurry!

Ron

On 22/11/2014 12:50 PM, Eric Wieling wrote:

[yves030 at gmx.de]

Hi,

I know this Bug,,, at least when you´re talking about x-lite 3... quite
annoying, but if you know it...
so no... its not the phone... tested with zoiper and 3cx ... both
work...but the problem occurs ONLY,
as soon as I register at more than one registrar...

yves

Am 22.11.2014 um 19:19 schrieb Ron Wheeler:

[yves030 at gmx.de]

Hi,

the useragents nothing to do with the problem... i tried numeric, alpha and alphanumeric... no difference.
they work all.... as long as I only use ONE registrar...
as soon as I register at more than one registrar... the line drops after 32 seconds.... really strange.

yves

Am 22.11.2014 um 19:01 schrieb Rafael Visser:

[marie at vtl.ee]

On 22.11.2014, at 13:40, Yves A. <yves030@gmx.de> wrote:

[amit at avhan.com]

Call drop after 30+sec happens if RTP is not received by asterisk for 30 seconds (RTP Timeout).
You should look for media IP address in SDP. If there is firewall, apart from port UDP/5060, you also need to open port UDP/10000-UDP/20000 (standard RTP ports)
You should try with RTP debug. It should show bidirectional traffic. If not, you surely have an issue with media IP or ports.

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Thanks & Regards,
Amit Patkar


On 11/27/2014 10:01 AM, Marie Fischer wrote: