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
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 Sangom
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
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
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 PROTEC
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,
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
___
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Aste
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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Asterisk-Users@lists.digium.com
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 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
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,
Ja
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 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 :
> 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.
2010/12/20 Paul Belanger :
> 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_
2010/12/21 Paul Belanger :
> 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 UT
Hi all,
I have strange problem - maybe someone could help.
Asterisk 1.6 running on Debian installed on HP DL380 with SmartArray,
2GB RAM, Xeon 3GHz on board.
Just after the call is answered all is just fine but several minutes
after that voice quality becomes poor.
Something like quantization eff
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 to
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 insecure=por
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
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