Hi Eric,
By saying "signalling" do you also mean a caller id signalling?
Thanks :)
On 6/2/12, Eric Wieling wrote:
> This is incorrect. The vast majority of settings in chan_dahdi.conf are
> applied when you do a module reload chan_dahdi.so
>
> You cannot change signaling, switchtype, or add or
I hope you understand restart as restarting asterisk service.
restart Asterisk (service asterisk restart) or from CLI -> restart
gracefully now (relevant command line)
and not rebooting the server.
chan_xtra also utilizes similar hooks for the GSM cards.
module unloading and loading is also a
Thanks, Mitul :)
What if we change the chan_extra.conf do we also need to restart the
server, can not only module reload chan_extra?
Thanks again :)
On 6/2/12, Mitul Limbani wrote:
> Any changes inside chan_dahdi requires asterisk restart.
>
> you can restart asterisk gracefully, where by asteri
This is incorrect. The vast majority of settings in chan_dahdi.conf are
applied when you do a module reload chan_dahdi.so
You cannot change signaling, switchtype, or add or remove channels (I'm sure
there are a few others) on a module reload, but most settings will be applied
on a reload.
If
Any changes inside chan_dahdi requires asterisk restart.
you can restart asterisk gracefully, where by asterisk will honor the
existing calls, but wont honor new calls till it restarts.
Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani
Ok,understand :)
Just now I change usecallerid=yes to usecallerid=no in chan_dahdi.conf
and do a module reload chan_dahdi but it still answer the incoming
call until 3-4 rings. Do I need to restart the server or module reload
is enough since I can't restart the server right now.
Thanks :)
On 6/2/
That depends on what country you are in. In the USA CallerID information is
sent between the first and 2nd ring.Asterisk defaults to expecting USA
style CallerID. If you are not in the USA then you'll have to research on how
to get CallerID working with Asterisk for your country. Search t
Thanks Eric for the prompt reply :)
Honestly I still need the caller id but I already strugle for around
1-2months to get the caller id work on my system :( yesterday I bought
a caller id converter hoping it will solve my problem but look like
it's not. I'm still trying to get the caller id to wor
Try usecallerid=no
The immediate= option is mainly for FXS ports and is almost never used.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta
Sent: Saturday, June 02, 2012 12:06 AM
To: Asterisk Us
- "dialplan show internal" shows the label (and if I change it, it changes to
the right one, etc.)
- No syntax errors or warnings showing in the CLI when I reload with verbosity
at 3.
- I'll check moving things around on Monday, but I find it odd that all I have
to do to make the GotoIf work w
On Wed, 30 May 2012 18:02:00 +
Noah Engelberth wrote:
> I have a hotdesking environment at my main office, and up until
> today, the GotoIf that jumps straight to voicemail if a user isn't
> log in was working just fine by label. Today, I deployed DUNDi to a
> satellite office, and now the G
- Original Message -
> Hi Tim,
> Thanks for your response. Here is my topology as listing down below;
> PSTN Line --> Cisco Voice GW --> IP Cloud --> Asterisk
> Will Asterisk able to receive the fax (as in topology above) using
> its' fax module? In sip.conf I enabled fax detection and
advice.
>
>
> --
> Regards,
>
> Ahmed Munir Chohan
> -- next part --
> An HTML attachment was scrubbed...
> URL: <
> http://lists.digium.com/pipermail/asterisk-users/attachments/20120601/3c861c56/attachment-0001.htm
> >
>
> --
All,
We are having issues with one of our customers. They typically are
using remote sip clients on smart phones. For the purpose of allowing
the apps to work properly in the background we have to utilize TCP which
works fine.
The problem comes up when the softphone loses connectivity
Ade-
> Does anyone do a low profile PCIe FXO card? I just picked up an HP ProLiant
> microserver for $nuppence, which I'd hoped to migrate my Asterisk setup
> onto. I currently use an A400P analog card, but the ProLiant only has PCIe
> slots, and they're short ones too, so I can't use an A400E car
Calling into 10.5.0-rc2 from a pstn did provider, I get no audio:
-- Executing [111@from-teliax:1] Dial("SIP/teliax-0010",
"SIP/office2/+1") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/office2/+1
-- SIP/office2-0011 answered SIP/t
Last time I checked (a few years ago) Sangoma has half height brackets
available. Contact their support or sales.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ade Vickers
Sent: Friday, June 01, 2012 10:41
- Original Message -
> Hi,
> Does anyone do a low profile PCIe FXO card? I just picked up an HP
> ProLiant microserver for $nuppence, which I'd hoped to migrate my
> Asterisk setup onto. I currently use an A400P analog card, but the
> ProLiant only has PCIe slots, and they're short ones to
Hi,
Does anyone do a low profile PCIe FXO card? I just picked up an HP ProLiant
microserver for $nuppence, which I'd hoped to migrate my Asterisk setup
onto. I currently use an A400P analog card, but the ProLiant only has PCIe
slots, and they're short ones too, so I can't use an A400E card. Even
- Original Message -
> Hi Tim,
> Unfortunately i can't reproduce the scenario because it was a long
> time ago. But it would be nice to hear from you, what things can be
> verified within fax and Asterisk? Any TIP on wireshark monitoring?
Within Asterisk, the debug logs can be helpful for
- Original Message -
> Hi all,
> Couple of things I would like ask, does Asterisk provides free
> license for FoIP (for 1 channel) or need to purchase it? Couple of
> years back, I was able to send and receive the fax using Digium T1
> card, in term of FoIP how can I able to receive fax fr
Hi all,
Couple of things I would like ask, does Asterisk provides free license for
FoIP (for 1 channel) or need to purchase it? Couple of years back, I was
able to send and receive the fax using Digium T1 card, in term of FoIP how
can I able to receive fax from traditional telephone lines / T1 lin
http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback
On Fri, Jun 1, 2012 at 1:45 PM, Satish Barot wrote:
> I believe you want your caller to request for a callback while he/she
> waits in a queue and when your agents are free, you want to call him back
> and place in a same position in a
I believe you want your caller to request for a callback while he/she waits
in a queue and when your agents are free, you want to call him back and
place in a same position in a Queue where he/she has left the Queue.
There exists an ugly(!) way of doing this.
(1)Set parameter 'context' in queues.
On 31.5.2012 г. 20:43 ч., Patrick Lists wrote:
Hi zoa,
On 31-05-12 17:39, joachim wrote:
Ellow,
We released zoiper for Android today, available for free here:
https://play.google.com/store/apps/details?id=com.zoiper.android.app
SIP and IAX is supported, should work quite well, unfortunately it
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