Hi,
I am facing issue while generating a dial plan for the following case:
all caller should be asked a code to enter than All the callers should be
connected one extension.
also tell me testing scenario :
I have pbx setup and currently I have soft phones to use as extension.
Currently I have
I am using Asterisk 1.4.36 with Realtime Voicemail from a MySQL database, and
whilst I have it all working, I am unable to find a way to customize the
content of the email that gets sent to a user when they receive a voicemail.
In the past I just edited it in the voicemail.conf file and made
Hi,
My set up is :
Asterisk with B410P in NT mode -cat5 straight cable
Another PBX in TE mode
Is the 100 Ohm terminator you can find on B410P boards, necessary when
connecting in NT mode to another PBX (set in TE mode) ?
Cheers
--
Hello my friends,
Previously, we developed fastagi application with Erlang to run on asterisk
1.4, it run very well.
But when we try to migrate this application to interface with asterisk
1.6.2.13( the same with 1.6.1 or 1.6.0),
We found the hangup issue there, the hangup event will delayed
Jigar Joshi wrote:
Currently I have created a dial plan using vdp I tried submitting it
here but I don't know how to extract text version for the same .
After Googling a bit, I found that VDP is Visual Dial Plan for
Asterisk. Neat little application, but I doubt you'll find many if any
Hello
I am working on TDM2400p. I am having some problems like:
when i connect my analog phone with the card there is no dial tone, but i
can dial any extension... but after that i can't hear any voice from my
receiver i have used different phone sets but still i cant communicate with
other
Hi,
How to properly re-configure dahdi, when for instance I want to change from
TE to NT mode ?
I'm planning the following operations :
/etc/init.d/asterisk stop
/etc/init.d/dahdi stop
rmmod dahdi
rm /etc/asterisk/dahdi-channels.conf
rm /etc/dahdi/system.conf
rm /etc/dahdi/modules
nano
Hi,
Have just been testing asterisk 1.8.0, 1.8.0-rc5 and a trunk version from about
half an hour ago.
IAX Friends (Zoiper Softphones) don't stay registered for more than a few
seconds they start out with status unknown and quickly become unreachable, I am
using realtime with postgresql, with
What happens without using Realtime ?
--
_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
On Mon, 18 Oct 2010, Jigar Joshi wrote:
@Gilles here are my requirement.can you please help me .
On Mon, 18 Oct 2010, Steve Edwards wrote:
Are you putting this out to bid or are you just too lazy to read ATFOT
(http://downloads.oreilly.com/books/9780596510480.pdf)?
On Sat, 23 Oct 2010,
Sorry forgot to add this into my initial email.
The same happens with phones configured in iax.conf and the Realtime database
table.
[Oct 23 16:49:52] ERROR[1220]: chan_iax2.c:8770 update_registry: Bad address
cast to IPv4 etc.
Nic.
From: asterisk-users-boun...@lists.digium.com
Hello,
I'm trying to use SipSak to check if my Asterisk server is
available/running with the following :
sipsak -vv -s sip:usern...@sip.domain.tld -c sip:usern...@sip.domain.tld
--password guessthis --hostname XX.XX.XX.63
The SIP OPTION is received by Asterisk as follow :
OPTIONS
My internal lan is small, 100mb, all wired. aastra phones.
sip show peers
...
142/... 10.10.10.42 D A 5060 OK (136 ms)
144/... 10.10.10.44 D A 5060 OK (138 ms)
145/... 10.10.10.45 D A 5060 OK (133 ms)
But pings are 1ms:
ping
sean darcy wrote:
Why are the sip latencies so high? And is it a problem? And if so, how
do I fix it?
I've noted that if I run DNS on the Asterisk sever, that my ms times
drop by almost 50%
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
It depends on the type of sip end point, and how long it takes to respond to a
SIP event.
For example if I connect a Cisco 7960 IP phone to my Asterisk server over the
LAN, I always see registration times of over 100ms.
But if I connect X-Lite I get registration times of under 10ms.
On Sat, Oct 23, 2010 at 11:53 AM, Nic Colledge n...@njcolledge.net wrote:
Sorry forgot to add this into my initial email.
The same happens with phones configured in iax.conf and the Realtime
database table.
https://issues.asterisk.org/view.php?id=18183
I was able to reproduce a problem with
On 10/23/2010 12:38 PM, Doug Lytle wrote:
sean darcy wrote:
Why are the sip latencies so high? And is it a problem? And if so, how
do I fix it?
I've noted that if I run DNS on the Asterisk sever, that my ms times
drop by almost 50%
Doug
I hadn't set any dns on the aastra's. I just did
Paul,
Thanks for your reply, I saw that issue on the tracker when I was trying to
find a solution earlier but it looked completely different so I didn't give it
much thought.
This happens on my install with both Realtime phones and those configured in
iax.conf
I have just tried the branch you
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Why are the sip latencies so high? And is it a problem? And if so, how
do I fix it?
Not a problem at all. Just a goofy Cisco thing. Polycom and Linksys and
Grandstream are all a lot lower, but Cisco has always been high. We've seen
that for 4-5 years and never had issues.
--
If you want to turn off the audio totally you can set audio to NO (it's
probably the 4th or 5th in list of Global settings). Otherway is to blank
the file responsible to play that file and keeping the settings intact.
However, there are numerous options to turn on and off the various
announcements
On Sat, Oct 23, 2010 at 1:48 PM, Nic Colledge n...@njcolledge.net wrote:
I have just tried the branch you suggested and the problem remains. It's
worse with qualify=yes but still happens (albeit less frequently) with
qualify=no.
Okay, just reproduced your issue and looking at the code now.
With previous Asterisk versions when running asterisk -r a welcome
message is displayed with the version. I just upgraded to 1.8 and
noticed it is not appearing. All I get is Verbosity is at least 3 and
the console prompt. I looked at main/asterisk.c and still see the
welcome message code. Any
Hello, this isn't an Asterisk specific problem but I don't know who
else to ask for help.
This is my setup, it oftens finds double NAT situations:
[Asterisk box] - [Firewall IPCop] -INTERNET- [Random Router] - [Softphone]
In certain situations, when two or more client softphones use the port
Some other people noticed that a few days ago. I think Paul was
looking at it...
~
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lath...@gmail.com
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On Sat, Oct 23, 2010 at 3:03 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
Okay, just reproduced your issue and looking at the code now. :)
Ok, think I fixed it. You can either apply this patch to 1.8.0, or
svn update the branch I'm working on. Feedback is welcome.
--
Paul Belanger |
Hello list,
I have been using Cepstral's 8KHz voices for my text-to-speech service for
some time now, and have been noticing that the voice quality is really poor,
doesn't matter what phrase I give it to convert. None of the other 8KHz
voices I have ever used were this bad. It doesn't seem good
Are you using app_swift or wav files?
On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Hello list,
I have been using Cepstral's 8KHz voices for my text-to-speech service for
some time now, and have been noticing that the voice quality is really poor,
doesn't
It's replying so its up :)
On 23 Oct 2010 17:32, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
I'm trying to use SipSak to check if my Asterisk server is
available/running with the following :
sipsak -vv -s sip:usern...@sip.domain.tld -c sip:usern...@sip.domain.tld
--password
I am using app_swift.
As a side note, demo on their website also generates sounds which at places
sounds like robotic.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-23 6:03 PM, Darren Sessions dmsessi...@gmail.com wrote:
Are you using app_swift or wav files?
On Sat, Oct 23, 2010 at 12:31 PM, sean darcy seandar...@gmail.com wrote:
My internal lan is small, 100mb, all wired. aastra phones.
sip show peers
...
142/... 10.10.10.42 D A 5060 OK (136 ms)
144/... 10.10.10.44 D A 5060 OK (138 ms)
145/...
Since it is Saturday evening (7PM EST) I am asking this on the list in case
someone who knows sees it and can answer.
Astricon is in my back yard for the first time, and I could hit you with a
rock. I would always like to attend, and spoke at the 2007 Astricon in
Phoenix but don't have the idle
On Sat, Oct 23, 2010 at 7:07 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
Question: Can I just go to Astricon and take the dCAP exam only? In and
out? Cost?
http://www.astricon.net/dCAP.aspx
Looks like cost is $300.
Is the cert on 1.4 or 1.6 now?
When I did it in May, I used
On Sat, Oct 23, 2010 at 3:35 PM, Andrew Latham lath...@gmail.com wrote:
Some other people noticed that a few days ago. I think Paul was
looking at it...
Here is the thread on asterisk-dev
http://lists.digium.com/pipermail/asterisk-dev/2010-October/046697.html
--
Paul Belanger | dCAP
On Sat, Oct 23, 2010 at 7:36 PM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Sat, Oct 23, 2010 at 7:07 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
Question: Can I just go to Astricon and take the dCAP exam only? In and
out? Cost?
http://www.astricon.net/dCAP.aspx
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