Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-28 Thread Stefan Schmidt
David Backeberg schrieb: On Wed, May 27, 2009 at 1:49 PM, Stefan Schmidt s...@sil.at wrote: all server are in one rack in our datacenter and are connected to an HP Procurve 2650 switch, which has been setup around 3 months ago, cause of the old switch died silent in the night. all server

Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-28 Thread Alex Samad
On Thu, May 28, 2009 at 10:49:38AM +0200, Stefan Schmidt wrote: David Backeberg schrieb: On Wed, May 27, 2009 at 1:49 PM, Stefan Schmidt s...@sil.at wrote: all server are in one rack in our datacenter and are connected to an HP Procurve 2650 switch, which has been setup around 3 months

Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-28 Thread Stefan Schmidt
Alex Samad schrieb: Hi Hi Alex, I am new to asterisk so my suggestions might be a bit silly. Why not setup a iax2 connection bettween the asterisk servers, because its a lower overhear and more efficient. We had changed from iax connections to sip connections cause we had timing

Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-28 Thread Elliot Otchet
-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan Schmidt Sent: Thursday, May 28, 2009 4:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stucked calls in asterisk 1.4 David Backeberg schrieb: On Wed, May 27

Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-28 Thread Alex Samad
On Thu, May 28, 2009 at 02:15:08PM +0200, Stefan Schmidt wrote: Alex Samad schrieb: Hi Hi Alex, I am new to asterisk so my suggestions might be a bit silly. Why not setup a iax2 connection bettween the asterisk servers, because its a lower overhear and more efficient. We

[asterisk-users] stucked calls in asterisk 1.4

2009-05-27 Thread Stefan Schmidt
hello, i have a problem with stucked or hanging calls in asterisk 1.4.25 we had this problem before and so we upgradet from 1.2.32 to 1.4.25 but it still exists and as i could see, happens even more. on this server there are 1500 clients registered all with qualify on and we had 2 routing

Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-27 Thread David Backeberg
On Wed, May 27, 2009 at 7:32 AM, Stefan Schmidt s...@sil.at wrote: i have a problem with stucked or hanging calls in asterisk 1.4.25 only appears on this server and not on the routing server. Even if i I'm confused. So the server where the calls get stuck has both SIP and DAHDI/Zaptel channel

Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-27 Thread Stefan Schmidt
hello David Backeberg schrieb: On Wed, May 27, 2009 at 7:32 AM, Stefan Schmidt s...@sil.at wrote: i have a problem with stucked or hanging calls in asterisk 1.4.25 only appears on this server and not on the routing server. Even if i I'm confused. So the server where the calls get stuck has

Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-27 Thread David Backeberg
On Wed, May 27, 2009 at 9:26 AM, Stefan Schmidt s...@sil.at wrote: Server A call it PBX there are the sip clients connected A call comes from server B or C to server A and then to a client, gets stucked on Server A when PSTN side hangs up. On server B or C the call You may not have properly

Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-27 Thread Stefan Schmidt
David Backeberg schrieb: On Wed, May 27, 2009 at 9:26 AM, Stefan Schmidt s...@sil.at wrote: Server A call it PBX there are the sip clients connected A call comes from server B or C to server A and then to a client, gets stucked on Server A when PSTN side hangs up. On server B or C the call

Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-27 Thread David Backeberg
On Wed, May 27, 2009 at 10:30 AM, Stefan Schmidt s...@sil.at wrote: as i said the routing server also handles calls from an ser proxy and another asterisk server where iax accounts terminates and this problem is only on the pbx server. Maybe it is a network problem but the quality of the rtp

Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-27 Thread Stefan Schmidt
David Backeberg schrieb: Now that I better understand your problem, I'm out of ideas. thats the point where i stand ;) You are correct that if a BYE sip packet gets lost, a) it won't get retransmitted if it's UDP b) the side that's waiting for the hangup will think the call is still active

Re: [asterisk-users] stucked calls in asterisk 1.4

2009-05-27 Thread David Backeberg
On Wed, May 27, 2009 at 1:49 PM, Stefan Schmidt s...@sil.at wrote: all server are in one rack in our datacenter and are connected to an HP Procurve 2650 switch, which has been setup around 3 months ago, cause of the old switch died silent in the night. all server had two interfaces and i have