Say, Is there any existing add-on / code etc. that manages speed dials.
I find myself dialing number repeatedly and think that it would be great to
have a system that can be controlled from the telephone instrument and work
on the fly to build up a speed dial list.
I would like that after I dial
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Hi!
If you haven't noticed yet, SER (the mother of the SIP proxy projects
Openser, Kamailio, sip-router, opensips, ) is celebrating their 10th
year. There will be a main event happening in Berlin
(http://sip-router.org/10-years-ser/).
For those who can not travel to this event, there will be
hello danny,
i don't have dahdi iuse zap
like below
exten = 0661760924,1,Set(CALLERID(number)=520460587)
exten = 0661760924,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten =
0661760924,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
exten =
this could be an unsupported codec. Do you know if Office365 supports PCMU? I
would also try to get rid of 101 (rfc2833) and see if that makes a difference
On Aug 15, 2011, at 8:40 PM, o o wrote:
Trying to make this work, and Office 365 support is useless, giving me the
following response
OK, I can buy echo canceller from Digium and how will be installed in the
digium card? Or it is a hardware?
Currently I am reading a message at the consol that Unable to enable the echo
canceller .. does this means that Digium card that I have is not supporting?
This is the output of the
Hi John,
I kind of facing the same problem that you were facing.
I am using similar configuration as you are for asterisk.
I am using java-asterisk library to communicate with asterisk.
In my code I am setting two variables (PIN, MREQID) and trying to access
them in dialplan (dialplan shown
Thanks for the help and reply.
And this can be done only by setting the callerid=5100 in the sip.conf? Or I
have to do any thing else?
Regards
Bilal
--
I need that if five IP Phones make outside calls, then
destination
should see only 56725111...
You can set the
Hi list,
I see from time to time this notice on asterisk CLI:
NOTICE[23601]: res_jabber.c:2051 aji_client_info_handler: *User Strange
Email/TalkGadgetX* does not support discovery.
The *Strange Email *is the email of an untrusted gay. I use Google talk with
Asterisk in my project.
What that
What that means?
Thank you for your help
On Tue, Aug 16, 2011 at 3:37 PM, A.H. Jos minustoplusinfin...@gmail.comwrote:
Hi list,
I see from time to time this notice on asterisk CLI:
NOTICE[23601]: res_jabber.c:2051 aji_client_info_handler: *User Strange
Email/TalkGadgetX* does not support
What I have done is created a special extension # (ie, 63XXX) and then
created a mysql database with XXX and the number to call, then when the
63xxx extension is dialed it looks up the number in the database via agi
script and completes the call.
From:
On Tue, Aug 16, 2011 at 4:42 AM, john Millican j...@millican.us
mailto:j...@millican.us wrote:
On 8/15/2011 5:48 PM, john Millican wrote:
Hello,
Asterisk 1.4.38
Linux version 2.6.9-89.31.1.EL CentOS
Trying to get variables into a dial plan from AMI.
I have seen this with last firmware.
You need to change those 2 parameters to get a non-autoreboot scenario:
Resync_At__HHmm_ ua=na0001/Resync_At__HHmm_
Resync_Periodic ua=na/Resync_Periodic
Whenever it resync profile, it reboots, so setting resync to an hour nobody
uses it and unseting
I am having a problem with a new PRI turn-up on dahdi 2.5.0 and
asterisk 1.8.5 that I have not seen before. The PRI is setup as B8ZS,
ESF and the span shows up and ok. This PRI is merely a crossover T1
going into an old DC0 class 5 switch.
I am getting the following errors over and over again
On Tue, Aug 16, 2011 at 05:34:58AM -0700, bilal ghayyad wrote:
OK, I can buy echo canceller from Digium and how will be installed in
the digium card? Or it is a hardware?
If you're using a TDM2400 you can buy a hardware echocan module or HPEC (which
is a proprietary software echocan) if you
On Tue, Aug 16, 2011 at 10:31:54AM -0400, Eric Merkel wrote:
I am having a problem with a new PRI turn-up on dahdi 2.5.0 and
asterisk 1.8.5 that I have not seen before. The PRI is setup as B8ZS,
ESF and the span shows up and ok. This PRI is merely a crossover T1
going into an old DC0 class 5
Agree -- make sure you are at the latest firmware.
ALSO: If you have provisioning enabled, and have a duplicate line in your
xml files, that will cause a reboot.
Cheers,
Cassius Smith
On 8/15/11 1:46 PM, C F shma...@gmail.com wrote:
I have 3 Linksys/Cisco 504G phones they keep restarting
Hi
I'm hoping someone could comment on how our setup will perform under
larger loads.
Its a quite simple setup, with Asterisk 1.6.2 on Debian 6 on an EC2 large
instance (7GB RAM, 2 virtual cores with EC2 compute units).
Using an IAX2 trunk we offer normal phones to dial in and listen to a mp3
As I understand it, the theoretical limits of Asterisk are hardware-based.
In real usage, I would see you running into some problems on C word boundary
limits (32665, etc). I have some 1.4 installs that run 2-5K calls a day
with minimal problems and I read frequently about users with 10K+ users.
On Tue, Aug 16, 2011 at 10:48 AM, Shaun Ruffell sruff...@digium.com wrote:
On Tue, Aug 16, 2011 at 10:31:54AM -0400, Eric Merkel wrote:
I am having a problem with a new PRI turn-up on dahdi 2.5.0 and
asterisk 1.8.5 that I have not seen before. The PRI is setup as B8ZS,
ESF and the span shows
Eric Merkel wrote:
The error pretty much right away.
The first thing that comes to my mind is to check your cable.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
Alex,
Thanks for the pointers. Digging through some Cisco documentation linked to
as a guide for configuring CCM 8.0 with Office 365, it states that they support
711ulaw . I also tried setting dtmfmode=auto/rfc2833/info/inband with no luck.
Trying to get someone with a brain at MS to work
does that mean you try setting dtmfmode=inband and made sure that 101 was no
longer present in SDP? Still you got 488?
good luck with that ;-)
On Aug 16, 2011, at 1:04 PM, o o wrote:
Alex,
Thanks for the pointers. Digging through some Cisco documentation linked
to as a guide for
Sorry for the top post, this is from my phone.
What you need to look at are the following:
Is it going to be just one mp3 stream that is shared across all users (I.e
everyone hears the same thing at the same time), or is it 1000 separate mp3
streams (everyone always starts at the beginning of
The current dahdi version is:
PBX-FF*CLI dahdi show version
DAHDI Version: 2.4.1.2 Echo Canceller:
Well, the output of the dahdi_cfg as shown below, it declares there is invalid
argument. But, really I tried to change the configuration in the systems.conf
from fxoks=1-16 to fxsks=1-16 but did
Hi Russ,
I have tried given patch and successfully compiled dahdi_pcap but when i try
to run below command it gives me error.
*./dahdi_pcap lapd 16 test.pcap *
error setting channel err=-1!
error setting channel err=-1!
error setting channel err=-1!
error setting channel err=-1!
Segmentation
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