Hi All,
I set up Homer SIPCapture and Captagent 6 on Asterisk box.
All works fine but SIP records are duplicated.
I tried to user extra filter "and not src host " into
captagent config but no success.
Can you point me how to figure it out?
Kind regards
Jarek
--
Hi all.
I have asterisk with sip registered accounts (realtime).
Moreover I have SIP trunk defined as type=peer in sip.conf.
When call is incoming from SIP trunk with CLID of one of sip friend defined
in MySQL sippeers table asterisk refuses INVITE as not authorized.
I tried to use
Hi all,
I have a Queue with 3 members:
SIP/100
SIP/200
SIP/300
When call arrives SIP/100 is ringing.. After given timeout ringing stops
but call is not routed to next member but SIP/100 starts ringing again.
I know that this is because SIP/100 is still available in the Queue but is
it any way
Hello,
I need to do such a simple thing:
1. Dial SIP/123
2. If I get for example 503 - jump to Dial SIP/789
3. If I get for example 403 - jump to Playback(...)
The real question is:
how can I get SIP Responses and use it in dialplan?
Regards,
Jarek
--
Hi,
I try to solve my problem with asterisk and BLF function.
I have registered peers from realtime with subscriptions but only type
is mwi (shown by 'sip show subscriptions').
Peers are registered from behind the NAT - may it be the cuase why
they not subscribed with dialog-info+xml?
Regards,
Hi all,
I try to figure out why I have empty :
sip show subscriptions
list in may asterisk 1.6.
When device is registering to asterisk I can see in log:
NOTICE[25603]: chan_sip.c:21518 handle_request_subscribe: Received SIP
subscribe for peer without mailbox: 1010
but
sip show subscriptions
Hi all,
I try to figure out why I have empty :
sip show subscriptions
list in may asterisk 1.6.
When device is registering to asterisk I can see in log:
NOTICE[25603]: chan_sip.c:21518 handle_request_subscribe: Received SIP
subscribe for peer without mailbox: 1010
but
sip show subscriptions
2010/12/21 Paul Belanger pabelan...@digium.com:
On 10-12-20 05:51 PM, Jarek Jarzebowski wrote:
OK, so I have attached debug log.
I am using:
*CLI core show version
Asterisk 1.8.1.1 built by root @ asterisk on a i686 running Linux on
2010-12-17 23:03:58 UTC
Definitely a bug, ran
Hi All,
I have some problem with Asterisk 1.8 and DIal() to SIP unreachable friend.
My dialplan:
exten = _,1,Dial(SIP/${EXTEN},60,rt)
Now, when I Dial extension 1050, and there is no 1050 peer registered I got:
[Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len
843) to
2010/12/20 Jeremy Kister asterisk...@jeremykister.com:
On 12/20/2010 4:41 AM, Jarek Jarzebowski wrote:
Now, when I Dial extension 1050, and there is no 1050 peer registered I
got:
[Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len
843) to 0.0.4.26:5060 returned -1
2010/12/20 Paul Belanger pabelan...@digium.com:
On 10-12-20 04:41 AM, Jarek Jarzebowski wrote:
[Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len
843) to 0.0.4.26:5060 returned -1: Invalid argument
It looks to be a regression with the IPv6 code added to chan_sip. Which
Dnia 31-12-2006 o 18:05:00 Jarek Jarzebowski [EMAIL PROTECTED] napisał(a):
Dnia 31-12-2006 o 17:39:10 Tzafrir Cohen [EMAIL PROTECTED]
napisał(a):
On Sun, Dec 31, 2006 at 05:08:26PM +0100, Jarek Jarzebowski wrote:
Dnia 31-12-2006 o 16:17:19 Tzafrir Cohen [EMAIL PROTECTED]
napisał
Hi All,
is anybody using Sangoma A102d card with Asterisk on Debian 3.1?
I configure and install Sangoma wanpipe step by step based on Sangoma Wiki
and manuals but can not get success results. I suppose that it may be some
Debian specific case.
Regards,
Jarek
Dnia 31-12-2006 o 16:17:19 Tzafrir Cohen [EMAIL PROTECTED]
napisał(a):
On Sun, Dec 31, 2006 at 03:59:14PM +0100, Jarek Jarzebowski wrote:
Hi All,
is anybody using Sangoma A102d card with Asterisk on Debian 3.1?
I configure and install Sangoma wanpipe step by step based on Sangoma
Wiki
Dnia 31-12-2006 o 17:19:35 Thomas Kenyon [EMAIL PROTECTED]
napisał(a):
Jarek Jarzebowski wrote:
Dnia 31-12-2006 o 16:17:19 Tzafrir Cohen [EMAIL PROTECTED]
napisał(a):
On Sun, Dec 31, 2006 at 03:59:14PM +0100, Jarek Jarzebowski wrote:
Hi All,
is anybody using Sangoma A102d card
Dnia 31-12-2006 o 17:31:18 Tzafrir Cohen [EMAIL PROTECTED]
napisał(a):
On Sun, Dec 31, 2006 at 04:19:35PM +, Thomas Kenyon wrote:
Jarek Jarzebowski wrote:
Dnia 31-12-2006 o 16:17:19 Tzafrir Cohen [EMAIL PROTECTED]
napisał(a):
On Sun, Dec 31, 2006 at 03:59:14PM +0100, Jarek Jarzebowski
Dnia 31-12-2006 o 17:39:10 Tzafrir Cohen [EMAIL PROTECTED]
napisał(a):
On Sun, Dec 31, 2006 at 05:08:26PM +0100, Jarek Jarzebowski wrote:
Dnia 31-12-2006 o 16:17:19 Tzafrir Cohen [EMAIL PROTECTED]
napisał(a):
On Sun, Dec 31, 2006 at 03:59:14PM +0100, Jarek Jarzebowski wrote:
Hi All
Hello,
is anybody there who successfully compiled Asterisk 1.2.4 with oh323 on
Debian Sarge? I tried severel versions of oh323 and pwlib and there is
no results... only errors.
--
Jarek
___
--Bandwidth and Colocation provided by Easynews.com --
Hi all,
I am trying to setup h.323 connection between two asterisks. The
situation is like that:
asterisk173 only must accept incomming h.323 calls from asterisk172, so
asterisk173 is peer and asterisk172 is user, am I right?
My config files:
Asterisk173:
ooh323.conf:
Hello,
I'm using Asterisk 1.2 with MySQL support. I use sip_buddies table for
SIP clients definition. My problem is that I can not define CLIR.
Sip.conf docs says that restrictid = yes hide caller identification. The
problem is that definition of sip_buddies field named restrictid is char(1).
Hi all,
I am new to asterisk and I can not find any detailed info on using SIP
MySQL support (sipfriends) with clients behind NAT. I've heard that I
have to patch chan_sip.c and Makefile to get it working.
I tried on voip-info.org but found no answer for my questions.
I found some answer on
Hi all,
I am interested in your opinions about using more then one Tormenta 2
card on asterisk server based on Debian - but distribution does not
matter in this case I suppose.
--
Jarek
___
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