Re: [asterisk-users] how to know RTP por of a SIP client in
Yes, I need to know to get in in dialplan because I want to capture traffic per
call. I would like to launch $SHELL{tcpdump src port } in the dialplan or
something like this. And I want RTP traffic only of a certain call.
Thank you!
===
Date: Fri, 21 Oct 2011 09:41:39 -0400
From: Bruce B bruceb...@gmail.com
Subject: Re: [asterisk-users] how to know RTP por of a SIP client in
the dialplan
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
CAJyE_uWLRXkhrWQ-6SvNW1ihN-nGA3HFwHt=pu-tfr6lybi...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Do you need to know to get it in dialplan? If I not, from shell (not
Asterisk CLI) I usually use:
netstata -a | grep asterisk
By default Asterisk settings it should be something between 10k-20k
-Bruce
On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Hi all,
How can I get the RTP port one SIP client is using for sending/receiving
RTP flow? Can I obtain it in from SIP_HEADER of something like that in the
dialplan?
Thank you!
** **
Isabel
Este mensaje se dirige exclusivamente a su destinatario. Puede consultar
nuestra política de envío y recepción de correo electrónico en el enlace
situado más abajo.
This message is intended exclusively for its addressee. We only send and
receive email on the basis of the terms set out at.
http://www.tid.es/ES/PAGINAS/disclaimer.aspx
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Peer and User Clarification
Hello All, It seems from the Asterisk documentation, a User places phone calls into the Asterisk server and a Peers accepts phone calls from the Asterisk server. However, according to the document describing the register = command for sip.conf, it seems that Peers can in fact place calls into an Asterisk system. Is this correct and how is this working? Thanks, Elliot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer and User Clarification
All endpoints are peers, in the broad sense of entities in sip.conf. This includes phones, gateways, provider endpoints, etc. When a phone makes a call through an Asterisk server, it initiates a call leg to Asterisk, which is matched to a sip.conf peer. Asterisk then initiates a second call leg through another sip.conf peer, and bridges the two legs together. Both are anchored by peers. The type of the peer (the type= setting) is a configuration detail that changes some minor aspects of how the endpoint is treated, but whether the type is friend, peer, etc. it's still a peer. They are largely the same. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Oct 23, 2011, at 5:46 AM, Elliot Murdock murdo...@gmail.com wrote: Hello All, It seems from the Asterisk documentation, a User places phone calls into the Asterisk server and a Peers accepts phone calls from the Asterisk server. However, according to the document describing the register = command for sip.conf, it seems that Peers can in fact place calls into an Asterisk system. Is this correct and how is this working? Thanks, Elliot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Questions on IAX client
Hi, I used to use Zoiper IAX to connect to my asterisk server from remote site. On asterisk CLI, I can see my zoiper client registered and stay on line. HOwever, I don't know why now I can't call this client. It always show up as Unable to create channel IAX2 (Cause 20 Unknown) I am using Asterisk 1.8.7.1 CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on IAX client
On Sun, 23 Oct 2011, asterisk asterisk wrote: I used to use Zoiper IAX to connect to my asterisk server from remote site. On asterisk CLI, I can see my zoiper client registered and stay on line. HOwever, I don't know why now I can't call this client. It always show up as Unable to create channel IAX2 (Cause 20 Unknown) If you enable IAX debugging you may get some clues. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running as non-root
On Wed, Oct 19, 2011 at 10:11:08AM -0400, David Backeberg wrote: If you use DAHDI, you need to change ownership of /dev/dahdi/* to the non-root owner. I ended up rolling that into the init script for dahdi. The init script of DAHDI or asterisk is the wrong place for that. If you're one of those who actually uses static files, you set their permissions at creation time or whenever. The rest of you: set the permissions in udev rules, as in the ones included with DAHDI. This avoids any potential races and unnecessary work. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running as non-root
On Thu, Oct 20, 2011 at 09:02:14AM +0200, Torbjörn Abrahamsson wrote: Thanks for all answers. One further question: If I run Asterisk as root, and set its group in asterisk.conf to apache, and make no changes to file/folder permissions, will I be able to run asterisk -rx 'clicmd' from a php-script (running as user apache with group apache)? 1. Why would you want to run Asterisk as root? 2. Why set the group to apache (if you're root anyway why do you care)? 3. If asterisk.ctl has the proper permissions, other users may issue some CLI commands, see cli_permissions.conf . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running as non-root
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: den 23 oktober 2011 21:19 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Running as non-root On Thu, Oct 20, 2011 at 09:02:14AM +0200, Torbjörn Abrahamsson wrote: Thanks for all answers. One further question: If I run Asterisk as root, and set its group in asterisk.conf to apache, and make no changes to file/folder permissions, will I be able to run asterisk -rx 'clicmd' from a php-script (running as user apache with group apache)? 1. Why would you want to run Asterisk as root? I wouldn't. But if my efforts to run asterisk as non root would hit a snag permission wise, this might be something I would consider. 2. Why set the group to apache (if you're root anyway why do you care)? It is not asterisk's permission to do things I am worried about. I want Apache to be able to issue asterisk cli-commands, like a reload after something crucial has changed in the web config. I am trying to ascertain if setting asterisk to run as group apache will be sufficient (actually regardless if asterisk runs as root or not), or if I still need to change permissions of files. In an other installation (this one made from RPMs, so non-root has been taken care of already) we have set apache to run as the same user as asterisk. 3. If asterisk.ctl has the proper permissions, other users may issue some CLI commands, see cli_permissions.conf . Will look at this, thanks! BR, Torbjörn Abrahamsson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running as non-root
On Sun, Oct 23, 2011 at 3:16 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Wed, Oct 19, 2011 at 10:11:08AM -0400, David Backeberg wrote: If you use DAHDI, you need to change ownership of /dev/dahdi/* to the non-root owner. I ended up rolling that into the init script for dahdi. The init script of DAHDI or asterisk is the wrong place for that. If you're one of those who actually uses static files, you set their permissions at creation time or whenever. The rest of you: set the permissions in udev rules, as in the ones included with DAHDI. This avoids any potential races and unnecessary work. Thanks for the tip. I just noticed that the permissions 'came undone' if I did a DAHDI reload, so it seemed like the right place. For the record, I'm also using SNMP with asterisk, also as non-root, and I'm also having a problem with /var/lib/masterx or whatever also reverting to being owned by root. And again, my presumptive fix is to put the chown directly into the SNMP script. Ideas? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to know RTP por of a SIP client in
Then you may use system() in dial-plan to run that shell command along with
what I suggested.
-Bruce
On Sun, Oct 23, 2011 at 5:22 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Yes, I need to know to get in in dialplan because I want to capture traffic
per call. I would like to launch $SHELL{tcpdump src port } in the
dialplan or something like this. And I want RTP traffic only of a certain
call.
Thank you!
===
Date: Fri, 21 Oct 2011 09:41:39 -0400
From: Bruce B bruceb...@gmail.com
Subject: Re: [asterisk-users] how to know RTP por of a SIP client in
the dialplan
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
CAJyE_uWLRXkhrWQ-6SvNW1ihN-nGA3HFwHt=pu-tfr6lybi...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Do you need to know to get it in dialplan? If I not, from shell (not
Asterisk CLI) I usually use:
netstata -a | grep asterisk
By default Asterisk settings it should be something between 10k-20k
-Bruce
On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Hi all,
How can I get the RTP port one SIP client is using for sending/receiving
RTP flow? Can I obtain it in from SIP_HEADER of something like that in
the
dialplan?
Thank you!
** **
Isabel
Este mensaje se dirige exclusivamente a su destinatario. Puede consultar
nuestra política de envío y recepción de correo electrónico en el enlace
situado más abajo.
This message is intended exclusively for its addressee. We only send and
receive email on the basis of the terms set out at.
http://www.tid.es/ES/PAGINAS/disclaimer.aspx
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
