Nathan,
Yes, SIP.. :-)
I ended up deciding to just not allow attended transfer at all since it
seemed so hard to deal with. If someone really wants attended transfer they
can put the call on hold, dial using the other line then transfer the call
on the other line if they want the call on the oth
Markus,
I'll take another shot at answering your questions. As before, if someone more
knowledgeable, like Joshua Colp, also responds please give more credibility to
their remarks.
> Although I have to say I don't understand what is going on exactly. :)
> As can be seen below, and as Joshua sug
On Wed, 17 Apr 2013, neo haux wrote:
I am using asterisk 11.1.0. How to display the caller number (from
asterisk -rvvv terminal) in the first step of the extension (before
doing any action) ?
Use 'verbose()' in priority 1. Note that this means whatever was at
priority 1 needs to be changed t
Hi,
I am using asterisk 11.1.0. How to display the caller number (from asterisk
-rvvv terminal) in the first step of the extension (before doing any
action) ?
Thanks
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I think facebook uses xmpp so you could use asterisk jabber or so
Don't know about the rest
-Original Message-
From: bilal ghayyad
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 17 Apr 2013 14:41:53
To:
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Joshua Colp wrote:
>
> Most of your response is correct except it doesn't take into account the
> rport RFC. Lack of implementation of an RFC doesn't make it
> non-compliant, so their stuff really is fine for this scenario. It all
> comes down to us forcing rport to be on by default.
>
> This
Hello;
Is there any modules or channels or integration between asterisk and any of the
following:
whatsapp, facebook, viber, yahoo and hotmail messanger?
Regards
Bilal
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I am using asterisk 1.8.5.0 and I am curious as to the roles of sip.conf
and users.conf.
My understanding is to provision phones you use users.conf. Doing so
creates a user, and a phone profile. With that said my understanding is
that sip.conf is the prefered method for creating sip accounts sin
> Hi all,
>
> I have console debugging enabled in logger.conf:
> console => notice,warning,error,debug
>
> Then a issue de command:
> core set debug 100 manager.c
>
> To see only debugging messages from AMI.
>
> But It shows nothing!!!
>
> And then if I do:
> core set debug 1
>
> Then I can s
Hi all,
I have console debugging enabled in logger.conf:
console => notice,warning,error,debug
Then a issue de command:
core set debug 100 manager.c
To see only debugging messages from AMI.
But It shows nothing!!!
And then if I do:
core set debug 1
Then I can see managar.c debug i
Is there a work around for Caller ID information not being persisted
when using the CLI or AMI Channel Redirect.
A calls B (caller id is displayed), B transfers call to C (no caller id
is displayed on phone c).
Jacob Miles
Software Engineer
jacob.e.mi...@l-3com.com
903.457.4422
Hello, We are using around 100 real time sip peers with phpagi. On asterisk
cli, getting frequent message 'failed to extend from 512 to 676'. Once we
execute 'sip reload', this message disappear for some time and then comes back.
Please let us know the solution for this. asterisk version 1.6.2.9
Matthew J. Roth wrote:
Joshua Colp wrote:
If you set nat=no for that specific peer it should work as you need.
'rport' is forced on these days which works for most situations, except
with some platforms and Cisco phones.>_>
Joshua,
That sounds much easier than what I came up with, so I'd reco
Hello List,
In PHPAGI, I'm using the Astrisk Manager function send_request() to
originate an outbound call. I want to execute the remaining PHP code after
the call gets executed (depending on user input). But presently the call
originates in a different context and asterisk executes the remaining
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