2012/10/11 Kinsey Moore kmo...@digium.com
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
Hi
I have just update my old asterisk 1.4 to 1.8 version and all works good
but I have a problem with the monitor features.
With the old version this part of dialplan worked without problem :
exten = _90.,1,NoOp(call out interoute REC)
exten = _90.,n,Monitor(wav,${CALLERID(num)}-${EXTEN:2}})
I made these changes in dialplan and it worked. Thanks a lot.
In most of the cases S1, S2 and C1 are in my control. But in some
cases the dialplan of C1 is not in my control. Also in some cases C1
can be any SIP client like a softphone or SIP device, so it wont work
in those case. Is there some
Deepesh D wrote:
I made these changes in dialplan and it worked. Thanks a lot.
In most of the cases S1, S2 and C1 are in my control. But in some
cases the dialplan of C1 is not in my control. Also in some cases C1
can be any SIP client like a softphone or SIP device, so it wont work
in those
I am using 1.4.43 connected on PRI to avaya PBX.
If I call one extension through the PRI and speak a message (recorded
file) sounds fine.
If I call an extension through the PRI that brings together a group of
phones on the avaya side
and play the same recorded file the audio drops out.
What
Any ideas?
On Thu, Oct 11, 2012 at 2:32 PM, Vik Killa vipki...@gmail.com wrote:
Call was to 7167436110
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This doesn't work reliably well with all all clients. I tested it
using a zoiper soft phone and it worked. But from an ATA device it
failed. On the S2 server it failed to authenticate
The console of S2 showed
[Oct 12 18:21:06] WARNING[30483]: chan_sip.c:13952 check_auth:
username mismatch, have
Deepesh D wrote:
This doesn't work reliably well with all all clients. I tested it
using a zoiper soft phone and it worked. But from an ATA device it
failed. On the S2 server it failed to authenticate
The console of S2 showed
[Oct 12 18:21:06] WARNING[30483]: chan_sip.c:13952 check_auth:
The trace is attached 3 emails back.
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Why am I feeling like I'm the only one here who is not able to see any
pastebin link or attachments in this thread !
On Fri, Oct 12, 2012 at 6:18 PM, Vik Killa vipki...@gmail.com wrote:
The trace is attached 3 emails back.
--
Sorry the attachment was too big. here is link:
http://www.2shared.com/file/Ola640Pn/doubledigit.html
On Fri, Oct 12, 2012 at 9:24 AM, SamyGo govoi...@gmail.com wrote:
Why am I feeling like I'm the only one here who is not able to see any
pastebin link or attachments in this thread !
--
On 10/12/2012 02:16 AM, Olivier wrote:
Yes I agree that new features should be committed to new versions but in
this specific case, current 1.8 behaviour is all Queue events but two
(ADDMEMBER and REMOVEMEMBER) are using members name for logging into
queue_log.
So, to me, queue_log
Last night we did a trial run. I am happy to report that both analog
and T1 lines worked well with the config files generated by
dahdi_genconf. Had a couple of minor issues that I'll ask about in
separate posts.
Of course when we got on-site, discovered that customer really has 6
analog
Setting up a group of analog lines to use for outbound emergency calls
(911). My current dial plan and debug output shown below. It appears
that when the SoftHangup() is executed that the line does not really
hang up. In the case shown, I had reduced the group to a single DAHDI
(analog)
Converting this customer from a MiTel system to asterisk. Discovered
that the inbound calls from the T1 are going to extension 366. (This
was mapped in the MiTel for some arcane purpose.) The dial plan I am
currently using is shown below. When loading the dial plan, I get this
warning:
On Fri, Oct 12, 2012 at 9:10 AM, Mitch Claborn mitch...@claborn.net wrote:
Converting this customer from a MiTel system to asterisk. Discovered that
the inbound calls from the T1 are going to extension 366. (This was mapped
in the MiTel for some arcane purpose.) The dial plan I am currently
On 10/11/2012 05:39 PM, Christopher Harrington wrote:
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
On Friday 12 October 2012, Mitch Claborn wrote:
Converting this customer from a MiTel system to asterisk. Discovered
that the inbound calls from the T1 are going to extension 366. (This
was mapped in the MiTel for some arcane purpose.) The dial plan I am
currently using is shown below. When
On Fri, Oct 12, 2012 at 9:59 AM, Matthew Jordan mjor...@digium.com wrote:
On 10/11/2012 05:39 PM, Christopher Harrington wrote:
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
On Thursday 11 October 2012, Christopher Harrington wrote:
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.
. [stuff deleted] .
From what I can tell, this appears to be a Digium phone limitation.
On Monday, October 15th, 2012, the Asterisk community services
listed below will be undergoing maintenance (software upgrades and
updates). The services will be shut down at approximately 9:00 PM CDT
(2:00 AM October 16th UTC), and will return no later than 10:00 PM
CDT. We apologize in advance
On 10/12/2012 10:35 AM, Christopher Harrington wrote:
On Fri, Oct 12, 2012 at 9:59 AM, Matthew Jordan mjor...@digium.com wrote:
Is type=peer strictly necessary? I don't know how they're currently
being specified from users.conf, is that possible to specify in
users.conf? I was under the
Setting up a group of analog lines to use for outbound emergency
calls
(911). My current dial plan and debug output shown below. It
appears
that when the SoftHangup() is executed that the line does not really
hang up. In the case shown, I had reduced the group to a single
DAHDI
(analog)
Setting up a group of analog lines to use for outbound emergency calls
(911). My current dial plan and debug output shown below. It appears
that when the SoftHangup() is executed that the line does not really
hang up. In the case shown, I had reduced the group to a single DAHDI
The s extension did not catch the incoming call. It was only when I
added a specific 366 or the _. wildcard that I was able to capture the
incoming call.
Mitch
On 10/12/2012 10:18 AM, A J Stiles wrote:
If (and only if) all the extensions you are using in all your contexts are
numeric,
On Fri, Oct 12, 2012 at 11:47 AM, Matthew Jordan mjor...@digium.com wrote:
After loading, [peer01] shows up as a known SIP endpoint. Calling
peer01 displays the same caller ID as before, i.e., either 101/D40 01 or
101/foo. Note that none of this using the DPMA either.
Here's something
Hi all,
I have an Asterisk PBX under development, that I would like to link to a
Skype account if possible. The idea is that people would call a particular
Skype username, and be redirected to my SIP and through that to Asterisk. Is
this doable? I have looked around and saw the Skype for
On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall phi...@blastbay.com wrote:
Hi all,
I have an Asterisk PBX under development, that I would like to link to a
Skype account if possible. The idea is that people would call a particular
Skype username, and be redirected to my SIP and through that
Hi all,
I configured the voicemail using realtime and for record voice messages,
I'm storing it in to MySQL DB as this setup works perfectly without any
issues. Later I tried to insert the custom greeting (busy) for VM in DB
for particular extension, it was unable to play the custom greeting but
On 13/10/2012, at 7:54 AM, Christopher Harrington ch...@acsdi.com wrote:
On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall phi...@blastbay.com wrote:
Hi all,
I have an Asterisk PBX under development, that I would like to link to a
Skype account if possible. The idea is that people would
From what I gather, it costs extra for each channel even for direct Skype to
Asterisk calls. Since my plan was to use this for business purposes, I'd
need at least something like 30 channels which would be way out of my
monthly budget unfortunately.
Kind regards,
Philip Bennefall
-
The only inexpensive way is to get siptosis but the developer has stopped
the support and upgrade unfortunately. I have been using it for two years
or more.
Excellent quality and works very well
On Sat, Oct 13, 2012 at 5:17 AM, Philip Bennefall phi...@blastbay.comwrote:
From what I gather, it
Hi,
Suppose I have the following in my AEL dialplan:
context incoming-1 {
_. = {
Set(GROUP()=1);
goto incoming|${EXTEN}|1;
}
};
context incoming-2 {
_. = {
Set(GROUP()=2);
goto incoming|${EXTEN}|1;
}
};
context incoming {
fax = {
Do stuff for incoming fax...
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