Hi All;
When the call coming via the E1 dahdi and I handle the call (as first step) by
exten = 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be
disconnected instead of queued.
But, when I handle the call (as first step) by playing any sound file and then
send for the
On 11/15/2011 04:56 AM, bilal ghayyad wrote:
Hi All;
When the call coming via the E1 dahdi and I handle the call (as first step) by
exten = 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be
disconnected instead of queued.
But, when I handle the call (as first step) by
Hi guys,
I want to ask if its possible to make calls using one SIP account,
The problem is like this : I have an iPhone app and I want all my users to call
the same extension which is virtual extension to my call center,
so the iPhone app will be using the same SIP account for all users
lets say
On 11/15/2011 07:28 AM, Faraj Khasib wrote:
Hi guys,
I want to ask if its possible to make calls using one SIP account,
The problem is like this : I have an iPhone app and I want all my users to call
the same extension which is virtual extension to my call center,
so the iPhone app will be
Hi
You can make the call from all. For that u need not to register but to
receive the call you need to register one and that can be done by any one
iphone app
On Nov 15, 2011 7:03 PM, Faraj Khasib fkha...@iconnecths.com wrote:
Hi guys,
I want to ask if its possible to make calls using one SIP
I have phone system and I am connecting Asterisk to it trunk.
Now I want my iphone users (clients ) to call my call center which is in phone
system by using the same SIP account
the user will call asterik with for example 6000 as account then the asterik
will forward the call via trunk to that
On Thu, Nov 10, 2011 at 03:02:43PM -0500, eherr wrote:
Unfortunately, I didn't compile with DON'T_OPTIMIZE. Would this render my
backtrace.txt completely useless or should I still submit?
Don't bother. It makes the issue more aparent, but has a very large
performance hit. In some cases it will
btw the call is one direction from clients to Call center
My question can be rephrased can I make call without registration to an
registered SIP account?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On
Hi folks,
How can I take advantage of a high-bandwidth CODEC, like G.722, for
internal communications at my site, but use G.711 (alaw/ulaw) for all
other outgoing calls? I need G.711 to support Inband DTMF signaling.
As my site has multiple locations that are tied together with IAX
That's one of the uses of the SIP_CODEC dialplan variable. Just set it in
the context or the sip.conf or users.conf. In your particular case, just
set up a specific context for the IAX calls
[iax-in]
Exten = _X.,1,Set(SIP_CODEC=G722)
Exten = _X.,n,answer()
-Original Message-
From:
Remove all other codec
On Nov 15, 2011 8:17 PM, Jaap Winius jwin...@umrk.nl wrote:
Hi folks,
How can I take advantage of a high-bandwidth CODEC, like G.722, for
internal communications at my site, but use G.711 (alaw/ulaw) for all other
outgoing calls? I need G.711 to support Inband DTMF
The variable for outbound is (SIP_CODEC_OUTBOUND=g722)
But I think asterisk will try to transcode then because the preferred codec on
the phone is ulaw or so
-Original Message-
From: Danny Nicholas da...@debsinc.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 15 Nov 2011
On Tue, 2011-11-15 at 16:38 +0200, Tzafrir Cohen wrote:
On Thu, Nov 10, 2011 at 03:02:43PM -0500, eherr wrote:
Unfortunately, I didn't compile with DON'T_OPTIMIZE. Would this render my
backtrace.txt completely useless or should I still submit?
Don't bother. It makes the issue more
Hi,
I have a setup with 5 remote offices, each having a Asterisk PBX.
I then have a central office, also with an Asterisk PBX.
The remote offices have 2 links to the central office, a large link,
and a smaller, but more reliable link.
Unfortunately, using IAX is not an option for me.
Can I use 2
can I make call without registration to an registered SIP account? --
Yes, you can but first you need to set allowguest=yes in sip.conf (makes ur
server insecure)
I guess you can put in same user/sip account in all iphones and like (in
x-lite) don't let the phones register to server rather set
IMO you can do this (I have a 1.4 client with 3 SIP trunks). Call-limit
(or whatever flavor of that is applicable to your version) will let you
control the flow across the trunks. The priority dialing would most likely
have to be accomplished via AGI dialing since you would have to know if (a)
Hi,
Can I use 2 SIP Trunks from each remote offices to the central site
and permit 2 simultaneous calls across the SIP trunk that passes over
the smaller line, and permit 10 simultaneous calls across the larger
link?
Yes.
I also wish to have priorities, so that more important calls are
Hi,
We're using it here. As Ido asked, is there an alternative way of
getting the SIP response in the event a Dial() fails?
Cheers,
Kingsley.
On Thu, 2011-08-18 at 07:42 -0500, Matthew Nicholson wrote:
Greetings,
Recently a performance regression in chan_sip was discovered in Asterisk
1.8.
I see on my CentOS systems that certain users for particular subsystems
have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74.
My two questions are:
1. Is there a list of these standard assignments somewhere? Googling did
not turn up anything for me.
2. Are there standard
On 11/15/2011 09:58 AM, Tony Mountifield wrote:
I see on my CentOS systems that certain users for particular subsystems
have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74.
My two questions are:
1. Is there a list of these standard assignments somewhere? Googling did
On Tue, 15 Nov 2011, Tony Mountifield wrote:
I see on my CentOS systems that certain users for particular subsystems
have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74.
My two questions are:
1. Is there a list of these standard assignments somewhere? Googling did
not turn
In article 4ec28e0b.20...@digium.com,
Jason Parker jpar...@digium.com wrote:
On 11/15/2011 09:58 AM, Tony Mountifield wrote:
I see on my CentOS systems that certain users for particular subsystems
have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74.
My two questions
On 11/15/2011 10:42 AM, Tony Mountifield wrote:
Yes, I was hoping to use such a system user and group for asterisk, which
would not conflict with any other system package I might install in the
future, by virtue of being reserved for asterisk.
There shouldn't be any conflict either way.
On Tue, Nov 15, 2011 at 04:42:05PM +, Tony Mountifield wrote:
But it sounds like it is distro-specific.
No, it's system-specific. Debian for example will assign UIDs out of the
relevant range based on the order in which packages are installed.
Just use the textual UID/GID values, not the
In article alpine.deb.2.00.151609440.26...@unicorn.drogon.net,
Gordon Henderson gordon+aster...@drogon.net wrote:
On Tue, 15 Nov 2011, Tony Mountifield wrote:
I see on my CentOS systems that certain users for particular subsystems
have standardised UIDs and GIDs. For example mysql=27,
In article 4ec296b9.8040...@digium.com,
Jason Parker jpar...@digium.com wrote:
On 11/15/2011 10:42 AM, Tony Mountifield wrote:
Yes, I was hoping to use such a system user and group for asterisk, which
would not conflict with any other system package I might install in the
future, by virtue
On Tue, 15 Nov 2011, Tony Mountifield wrote:
In article 4ec296b9.8040...@digium.com,
Jason Parker jpar...@digium.com wrote:
On 11/15/2011 10:42 AM, Tony Mountifield wrote:
Yes, I was hoping to use such a system user and group for asterisk, which
would not conflict with any other system
On Tue, Nov 15, 2011 at 4:56 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
When the call coming via the E1 dahdi and I handle the call (as first
step) by exten = 5631040,1,Goto(OrangeCMG,s,1) it is not working and the
call will be disconnected instead of queued.
But, when I handle
Because playback is forcing an answer() before it starts; goto does not (no
implied media need).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Tuesday, November 15, 2011 2:39 PM
To: Asterisk Users Mailing List -
Thanks for the answer Danny. Can you give an example of when we would setup
peers through Method 1 2 as you described?
If I am using FreePBX setup generic SIP extensions then use Polycom phones
configure them to register with the SIP server (asterisk) with the
extension/user password, are
OK. Thanks everyone for the responses. If I can summarize, I think here's
what's been discussed:
Asterisk becomes aware of SIP extensions/peers, as soon as they register.
Regarding how asterisk becomes aware of (or determines) that they are
unavailable/unreachable, I believe I am hearing two
Hello,
Is there a way Asterisk can be used to send out SIP invites to
external Network Gateways? I.E., I have an Asterisk with some
softphones registered on it. I simply want to send out SIP invite, as
simple as sip:call...@domain.com;transport=tcp, to an external
Gateway, which will in turn
Hi
Thats is also one of the reason
On Tue, Nov 15, 2011 at 20:27, isr...@gmail.com wrote:
The variable for outbound is (SIP_CODEC_OUTBOUND=g722)
But I think asterisk will try to transcode then because the preferred
codec on the phone is ulaw or so
-Original Message-
From: Danny
Hi
This can be done.
On Wed, Nov 16, 2011 at 10:36, Amar Akshat amar.aks...@gmail.com wrote:
Hello,
Is there a way Asterisk can be used to send out SIP invites to
external Network Gateways? I.E., I have an Asterisk with some
softphones registered on it. I simply want to send out SIP
Hey,
I haven't thoroughly read the whole of your reply- just a quick answer to
your timers question-generally I think you're right. Those timers are
property of UAC so you may need to look into the phone configurations.
I'd CISCO 79X0 phones and we wanted those to refresh their registrations at
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