From C1 when I directly dial into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is
From C1 when I directly dial into S2, it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2, I define the context as
'test_context' and the default context is
From C1 when I directly dial into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is
On Wed, 2012-10-10 at 18:09 -0300, Joshua Colp wrote:
[snip]
Yes, there is no capability for video transcoding in any version of
Asterisk.
Thanks for pointing out!
So in case my managers starts nagging about it, they have two options:
A) use hard/soft-clients with comparable codecs,
B)
From C1 when I directly dial into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is
Hi,
Watching my queue_log file content, I can read entries like :
1349940957|1349940957.4|6025|Local/6455@from-queue/n|ADDMEMBER|
My extensions.conf file used this statement:
exten = *9876,n,AddQueueMember(6025,Local/6455@from-queue
/n,10,,FOOBAR,hint:6455@ext-local)
I was expecting to see:
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is 'default'.
When I directly dial from C1 into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default
Only callers calling from Earthlink internet connection
On Wed, Oct 10, 2012 at 5:18 PM, Don Kelly d...@donkelly.biz wrote:
Is this happening for all callers, or just iPhone callers?
--Don
--
_
-- Bandwidth and Colocation
Can you share your pcap trace !
On Thu, Oct 11, 2012 at 5:16 PM, Vik Killa vipki...@gmail.com wrote:
Only callers calling from Earthlink internet connection
On Wed, Oct 10, 2012 at 5:18 PM, Don Kelly d...@donkelly.biz wrote:
Is this happening for all callers, or just iPhone callers?
So what happens when you dial directly from S1 to S2?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D
Sent: Thursday, October 11, 2012 5:44 AM
To: Asterisk Users Mailing List - Non-Commercial
How do I disable IAX2 caching or provisioning features to eliminate this
repeated message:
ERROR[2493]: iax2-provision.c:266 iax_provision_version:
--
Joseph
--
_
-- Bandwidth and Colocation Provided by
I am able to register S1 as a peer in S2 and dial from S1 to S2, but
this is not my requirement. I want to dial from C1 into S1 and S1
should redirect the call to S2.
I am trying to do a load balancing setup between S1 and S2. S1 will be
primary server which accepts all calls and then based on
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D
Sent: Thursday, October 11, 2012 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to use 'Transfer'
On 10/11/2012 08:16 AM, Joseph wrote:
How do I disable IAX2 caching or provisioning features to eliminate
this repeated message:
ERROR[2493]: iax2-provision.c:266 iax_provision_version:
Hi Joseph:
I believe your issue was reported in ASTERISK-20337, and should be fixed
in the current 1.8
Have you tried Dial instead of Transfer?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D
Sent: Thursday, October 11, 2012 2:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
In case the moderator doesn't approve my post with the attachment, below
is a quick and dirty transcription of the order form.
Customer connecting equipment: CSU/DSU
Circuit: DS1
Line coding: B8ZS
Framing: ESF
Jack type: RJ48X / Smartjack (will that fit into the Digum card?)
ISDN protocol: NI2
Hi Olivier,
My questions are:
1. Is it possible to configure Asterisk 1.8 (I'm using 1.8.12 but I can
upgrade to 1.8.17 if, and only if, necessary) so that ADDMEMBER entries in
queue_log refers to member name instead of member location ?
If positive how should this be configured ?
This feature
On 10/11/2012 09:34 AM, Mitch Claborn wrote:
In case the moderator doesn't approve my post with the attachment,
below is a quick and dirty transcription of the order form.
Customer connecting equipment: CSU/DSU
Circuit: DS1
Line coding: B8ZS
Framing: ESF
Jack type: RJ48X / Smartjack (will that
2012/10/11 Kinsey Moore kmo...@digium.com
Hi Olivier,
My questions are:
1. Is it possible to configure Asterisk 1.8 (I'm using 1.8.12 but I can
upgrade to 1.8.17 if, and only if, necessary) so that ADDMEMBER entries in
queue_log refers to member name instead of member location ?
If positive
I've tested asterisk 1.8.17.0 and I'm still getting the repeated
error message on the command line:
iax2-provision.c:266 iax_provision_version: ast_db_get failed to
retrieve iax/provisioning/cach
Are you out of disk space? I would only expect to see that message once
since it looks like
I've tested asterisk 1.8.17.0 and I'm still getting the repeated
error message on the command line:
iax2-provision.c:266 iax_provision_version: ast_db_get failed to
retrieve iax/provisioning/cach
Are you out of disk space? I would only expect to see that message
once
since it
On 10/11/2012 10:31 AM, Olivier wrote:
2012/10/11 Kinsey Moore kmo...@digium.com mailto:kmo...@digium.com
Hi Olivier,
My questions are:
1. Is it possible to configure Asterisk 1.8 (I'm using 1.8.12 but
I can
upgrade to 1.8.17 if, and only if, necessary) so that ADDMEMBER
Hi,
I've been trying to route incoming calls based on CID to a trunk but the
calls are not getting though. I am trying to use a wild card prefix based
on countries so I can point the call to the appropriate trunk.
I am running Asterisk 1.8 with FreePBX.
Here is a sample of my configuration
If C1 dials S1 and then S1 dials S2 to transfer the call then S1 still
remains in the loop till the call is finished. What I wanted to do is
to reduce the number of calls on S1, so as soon as S1 receives a call
from C1 it redirects the call to S2 using 'Transfer' application and
exits from the
Well we trashed both 11 installs (11 from tgz on site and 11 from svn) as
the configuration just wouldn't work.
Again reverting to Asterisk 10 with NO network changes / no machine /
firewall changes worked instantly.
We also threw wireshark up and saw no rtp or other such audio path when on
11.
Try: exten = _00336123412xx/_44XX.,1,Set(RINGTIME=90,g)
Notice the _ on your callerid pattern
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoffrey Yeoh
Sent: Thursday, October 11, 2012 1:15 PM
To:
Robert wrote:
Well we trashed both 11 installs (11 from tgz on site and 11 from svn) as
the configuration just wouldn't work.
Again reverting to Asterisk 10 with NO network changes / no machine /
firewall changes worked instantly.
We also threw wireshark up and saw no rtp or other such audio
Has anyone seen issues with recent Sangoma T-1 cards and Sangoma Analog cards
on multiple different servers?
On T-1: we get NO traffic, no interrupts, and no increase in number of packets
and the PRI does not come up.
On Analog: The ports do NOT go red when you unplug the phone line from FXO
Deepesh D wrote:
If C1 dials S1 and then S1 dials S2 to transfer the call then S1 still
remains in the loop till the call is finished. What I wanted to do is
to reduce the number of calls on S1, so as soon as S1 receives a call
from C1 it redirects the call to S2 using 'Transfer' application and
- Original Message -
Has anyone seen issues with recent Sangoma T-1 cards and Sangoma
Analog cards on multiple different servers?
On T-1: we get NO traffic, no interrupts, and no increase in number
of packets and the PRI does not come up.
On Analog: The ports do NOT go red when
Thanks Eric. That works.
--
Try: exten = _00336123412xx/_44XX.,1,Set(RINGTIME=90,g)
Notice the _ on your callerid pattern
-Original Message-
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
First post to this mailing list. I'll keep it brief: My D40 phones
don't show the name component of CALLERID.
It only displays the number. This includes calls originating from PSTN
with their own CID already set, and calls
where we've specifically filled in this data. Changing the destination
of
Hi.
I have a problem with the Caller ID in Ukraine - it is not coming, but I
sure that telco is providing it as a DTMF. I have recorded the line
using dahdi_monitor: http://dl.dropbox.com/u/2962/dtmf.wav
The record contains DTMF codes, first ring and answer of dialplan -- it
is just
Alexander Tarasov
Sent: Friday, 12 October 2012 12:10 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Caller ID DTMF is not coming
Hi.
I have a problem with the Caller ID in Ukraine - it is not
coming, but I sure that telco is providing it as a DTMF. I
have
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