Re: [asterisk-users] SIP RTP ports not released when channel is hung up
Hi, did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It looks related to your issue. Best regards, Marcus On Fri, Feb 12, 2010 at 12:04 PM, Armin Schindler ar...@melware.de wrote: On Fri, 12 Feb 2010, Armin Schindler wrote: I had a look at netstat -nuap and it shows that a lot of ports are still assigned, even if there is no channel in use. But sip show channels show a lot of (unused) entries with no codec/Format and Last Message like INVITE, REGISTER, OPTIONS. REGISTER and OPTIONS allocate no RTP ports, so those are not a problem. If you have a SIP channel that has a last message being INVITE and still say you have no calls, you have a problem right there. I just see these entries with sip show channels, but cannot tell if e.g. the REGISTER listed channels have RTP ports allocated. Who can I find out which SIP channel allocated which port? Or which SIP channel belongs to the ports I see with 'netstat -nuap'? I just made a test to confirm: After a restart of asterisk (to have a clean state with no sip channels activ and no RTP port allocated), I can confirm that: - REGISTER and OPTION listed sip channels don't use RTP ports - after some calls (e.g. SIP to SIP) the RTP ports are freed immediately (looks like this is the case on hangup before answer). - after some other calls, the RTP ports are freed after about 20-30 seconds after hangup. So basically all is correct. I do have a sip channels like 172.21.4.1146660430c3a638e 00102/0 0x0 (nothing)No Init: INVITE in 'sip show channels' and they don't go away for a long time. Shouldn't there be a timeout to destroy such a channel even if somehow the phone was 'disconnected' in during a call? If the channels exists even after 32 seconds after BYE, and BYE was signaled correctly, I would file a bug report. It really looks like that there is a case where the sip channel is not destroyed and that is the cause of the problem. I will try to reproduce this. Any ideas? Armin -- _ -- 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 -- Dipl.-Inf. (FH) Marcus Hunger - hun...@sipgate.de Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106 / 5724 / 7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk -- _ -- 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
Re: [asterisk-users] Audio issue in skype for asterisk
Hi, we have a similar problem. When we try to make two skype-calls at a time, only one of them has working audio. For this to happen, both calls must be ringing at the same time. Does anyone know how to fix this? Best regards, Marcus Hunger On Thu, Oct 22, 2009 at 10:45 AM, Samir Doshi smrdo...@gmail.com wrote: Hi, I am facing audio issue in my skype for asterisk setup. *Flow of the call is like this.* e.g. Skype users : test2 Sip users: 1001 1002 -- test2 This both sip users 1001 and 1002 are register in same asterisk. And also test2 skype user is register in same asterisk. Now 1001 is dialing skype user test2 (skypeout). So, test2 is getting call. But as test2 skype user is register in our asterisk, our asterisk is getting that call (skypein). And test2 is mapped with 1002 user. So when test2 user call comes to asterisk our asterisk is dialing SIP/1002. And 1002 is getting calls. But when 1001 and 1002 user is connecting they are not getting audio. But this is working fine for only skypout and skypein. But when call come back to asterisk audio issue is coming. I have checked rtp debug, But getting proper packages in rtp debug. I am attaching image of call flow. [image: ?ui=2view=attth=1247ba299ad2be9dattid=0.1disp=attdrealattid=ii_1247ba299ad2be9dzw] Please help me to fix the issue. -- Thanks, Samir Doshi ___ -- 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 -- Dipl.-Inf. (FH) Marcus Hunger - hun...@sipgate.de Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106 / 5724 / 7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk skypeforasterisk.jpeg___ -- 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
Re: [asterisk-users] evaluate SIP response codes in dialplan
Hi, sorry for joining the discussion so lately. I'd like to ask you to check http://bugs.digium.com/view.php?id=14810. The patch tries to address the issue using channel-variables to propagate the hangup-cause to the calling channel. Best regards, Marcus On Fri, Jan 23, 2009 at 3:08 PM, Johansson Olle E o...@edvina.net wrote: 21 jan 2009 kl. 11.49 skrev Klaus Darilion: Hi Olle! Currently we have the problem that due to SIP-hangupcause-SIP-hangupcause conversions the original hangupcause gets lost in a chain of Asterisk servers using SIP. In chan_sip there is already code for adding the X-Asterisk-Hangupcode header. What about reading this header on the receiving side for setting the hangupcause instead of doing SIP-hangupcause mapping ? In this case we could do that, but there has to be an option to enable it since it will change the behaviour in existing networks. Good idea! /O ___ -- 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 -- Dipl.-Inf. (FH) Marcus Hunger - hun...@sipgate.de Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106 / 5724 / 7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk ___ -- 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
[asterisk-users] sip.conf setvar option
Hi, does anybody know about the setvar option in asterisk's sip.conf. I am trying to define it for a peer that's used when making calls using the originate ami call, but it seems to not have any effect. Marcus -- Marcus Hunger - [EMAIL PROTECTED] Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 indigo networks GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5713/2881, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk ___ -- 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
Re: [asterisk-users] sip.conf setvar option
So, wouldn't it be great to enable setvar for outgoing calls too?
On Fri, Mar 28, 2008 at 1:55 PM, Johansson Olle E [EMAIL PROTECTED] wrote:
28 mar 2008 kl. 13.42 skrev Jared Smith:
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in asterisk's sip.conf.
Sure! This is one of my favorite features.
Let's say I have a definition for my phone in sip.conf, and it looks
something like this:
[myphone]
secret=verysecretpassword
type=friend ; a friend is both a user and a peer
host=dynamic ; phone will register to Asterisk
disallow=all
allow=gsm ; first, try to negotiate gsm
allow=ulaw; the try ulaw
setvar=MYVAR=blah
Whenever a call comes into Asterisk from this particular phone,
Asterisk
will automatically create a channel variable named MYVAR, and ${MYVAR}
will contain the value blah. I can then use it for whatever
purpose I
see fit within my dialplan.
Well, Jared, but that's the reverse. You stripped out this important
part:
am trying to define it for a peer that's used when making calls
using the originate ami call, but it seems to not have any effect.
The important thing with your lesson was that SETVAR is only used on
INCOMING calls from
devices, not outbound calls TO devices. Using ORIGINATE to call a SIP
peer, there's no variables
set from sip.conf.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/ * SIP Masterclass
Orlando FL * April 21-25 2008
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--
Marcus Hunger - [EMAIL PROTECTED]
Telefon: +49 (0)211-63 55 55-61
Telefax: +49 (0)211-63 55 55-22
indigo networks GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5713/2881, Umsatzsteuer-ID: DE219349391
www.sipgate.de - www.sipgate.at - www.sipgate.co.uk
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Re: [asterisk-users] sip.conf setvar option
Particularly, I want to set the SIPADDHEADER variable dynamicly for peers with rt-engine. Working around it might be possible, but having the thing working transparently for Dial and Originate would be great. On Fri, Mar 28, 2008 at 2:47 PM, Johansson Olle E [EMAIL PROTECTED] wrote: 28 mar 2008 kl. 14.00 skrev Marcus Hunger: So, wouldn't it be great to enable setvar for outgoing calls too? Well, maybe in the outbound channel then. But that won't help much. mixing the caller's and callee's variables in the INCOMING channel would be messy and only cause issues. But there's another way. Hint hint. Friday afternoon hack. /O ;-) ___ -- 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 -- Marcus Hunger - [EMAIL PROTECTED] Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 indigo networks GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5713/2881, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk ___ -- 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
