Hello
We are experiencing some difficulties with T38 faxing.
I have a Asterisk 11.5.0 with libss7 and Sangoma A104DE digital interface card
. The operating system is Centos 6
We are using this server to terminate calls to Telco.
So calls are coming to asterisk from sip and we are sending calls
On 31/08/14 17:40, Marie Fischer wrote:
Le 29 août 2014 14:30, Marie Fischer ma...@vtl.ee a écrit :
Hi,
we use OSX CardDAV server and its response is very slow, so we
ended up syncing all the CardDAV contacts to MySQL via cron.
Asterisk dialplan then runs a query defined in
Hi All,
I would like to Setup own IP PBX Server for our office.
I need to connect our all branch office with head quarter through local
extensions.
I need to receive and make call from our branch office and head quarter
using own DID numbers.
Our branch offices located in India,Indonesia,
On Monday 01 Sep 2014, Chandran Manikandan wrote:
Hi All,
I would like to Setup own IP PBX Server for our office.
I need to connect our all branch office with head quarter through local
extensions.
I need to receive and make call from our branch office and head quarter
using own DID numbers.
On 01.09.2014, at 11:42, Lukasz Sokol el.es...@gmail.com wrote:
On 31/08/14 17:40, Marie Fischer wrote:
Le 29 août 2014 14:30, Marie Fischer ma...@vtl.ee a écrit :
we use OSX CardDAV server and its response is very slow, so we
ended up syncing all the CardDAV contacts to MySQL via cron.
Hi All
am running asterisk 1.8 , its a realtime asterisk.have recently noticed an
issues where am getting loads of these errors
SIP/blabla requested media update control 26, passing it to SIP/ProviderX
this literally keeps flooding the screen untill the call is answered and then i
get normal
Hey everybody.
Another XMPP+Asterisk example:
http://www.mundoopensource.com.br/en_page_send-xmpp-message-extensions-logged-asterisk-queue/
[]s
Marcelo H. Terres
mhter...@gmail.com
Openfire-BR mailing list's owner
IM: marc...@jabber.mundoopensource.com.br
http://www.mundoopensource.com.br
Hi Guys,
Do you know of any asterisk community version that does video codec
trans-coding or in other words supports video? I have 1.8.8.1 and I see
h263.c format files but can't see that codec in make menuselect. it might
be just a license issue (if h263 has to have license), but not sure if
Khalid Touati wrote:
Hi Guys,
Kia ora,
Do you know of any asterisk community version that does video codec
trans-coding or in other words supports video? I have 1.8.8.1 and I see
h263.c format files but can't see that codec in make menuselect. it
might be just a license issue (if h263 has to
Hello,
I have two sip phones (zoiper). Earlier these used to communicate using the
settings below for sip.conf and extensions.conf and now we asterisk
1.8.29.0, so these phones have stopped communicating. My question is that
does 1.8.29.0 release require any more changes to be done to the
what do you get on the asterisk console output ?
Date: Mon, 1 Sep 2014 18:53:51 +0530
From: dee...@voxomos.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Calls Not Working
Hello,
I have two sip phones (zoiper). Earlier these used to
communicate using the settings below
== Using SIP RTP CoS mark 5
-- Executing [100@exten-101:1] Dial(SIP/101-0014, SIP/100) in new
stack
== Using SIP RTP CoS mark 5
-- Called SIP/100
-- Registered SIP '101' at 115.252.66.70:55258
[Sep 1 18:10:20] NOTICE[4629]: chan_sip.c:25735 handle_request_subscribe:
Received SIP subscribe for
the warning message
[Sep 1 18:10:43] WARNING[4629]: chan_sip.c:3982 retrans_pkt: Retransmission
timeout reached on transmission 5f0235b842799d285a70eb2d452974fb@dynamic for
seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/ ...
nsmissions
Packet timed out after 32000ms
Any article that goes through this (seems to be tedious) task to add video
support and patents?
On Mon, Sep 1, 2014 at 8:14 AM, Joshua Colp jc...@digium.com wrote:
Khalid Touati wrote:
Hi Guys,
Kia ora,
Do you know of any asterisk community version that does video codec
trans-coding
Hello,
On a Asterisk 11.12.0, I'm studying BLF behaviour with Yealink phones.
My ultimate goal is to present Operator the name and number of every
incoming call so that he/she can if it's worth to pickup a ringing
incoming call.
I've discovered notifycid option in sip.conf.
When a call comes
Are there any known complications? I believe this should br straight forward.
IT ENGINEER
Original message
From: Olivier oza.4...@gmail.com
Date:09/01/2014 10:28 AM (GMT-06:00)
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sorry to bump such an old post. Which hub is that?
--
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New to Asterisk? Join us for a live introductory webinar every Thurs:
Hi
For connecting SIP phone users through PRI to local exchange we need
redundancy.
It is possible using 2 servers each server have 2 instance of Asterisk (totaly
4).
one instance in each server works in load share with instance running on
other server converting E1 PRI to SIP Trunk and
Hi Stiles,
Thanks for your reply. Please advise how to proceed to get response. I want
to get idea to setup our own IPBX system. So if any one willing to help me
it's great.
On Mon, Sep 1, 2014 at 6:43 PM, A J Stiles asterisk_l...@earthshod.co.uk
wrote:
On Monday 01 Sep 2014, Chandran
Hey All
We have several AGI scripts that access databases. These work well most of
the time.
The issue we are having is that on rare occasion our script must fail to a
backup database server.
When this occurs it may take up to two seconds to do so. The issue is
when there is this
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