Hmm... dunno about LCDial... *shrugs* We have multiple servers set up,
and the failover server is always in the dialplan after the primary...
Actually, our servers don't even have access to the dns for making
calls, only the phones, as our dns server died and screwed up all our
phone calls, so
I had very mysterious hangups of PSTN outbound calls which I traced
to zapata.conf. I had increased rx_gain = 10 and tx_gain = 10
because volume was always low because Polycom phone was always
resetting the volume. Anyway, I patched my sip.cfg for Polycom so my
volume would stay at the
Hi,I'm struggling to set up a plan to allow international dialing from my US location. As I understand it, an international call can have 9 to 15 digits including country code. The problem is that the call always goes through after I've entered the 9th digit.
My service provider is BroadVoice, my
I received an e-mail from a vendor who says:
We have recently become aware of an issue in the chan_iax2
implementation of IAX2. This issue leads to degraded audio quality.
Due to this we are urging everyone to move to SIP.
I don't want to discount what this person is talling me, but I'm
curious
One thing that may help:
I use outlook rule to move all the messages into a folder.
Then outlook has a feature, instead of sorting by date, or subject,
you
can sort by conservation.
Me too. Except, for some reason it often misses the first message of the
conversation (especially if I'm
On Mar 20, 2006, at 3:47 PM, Matt wrote:
I received an e-mail from a vendor who says:
We have recently become aware of an issue in the chan_iax2
implementation of IAX2. This issue leads to degraded audio quality.
Due to this we are urging everyone to move to SIP.
I don't want to discount
On Monday 20 March 2006 13:49, John Daragon wrote:
Hell, you learn something new every short period of time. I have to
go try this out...
:-) It's called early audio in PRI parlance, some carriers do not offer
it but almost all do.
Always pisses me off when I call AmEx at 1-800-297-1000
Hey group,
I have a Polycom 501 and a 301 together in my office. Each phone is
registered to a different server. When I call one of the phones from the
other, the other phone rings no problem (the calls are passed between
servers via IAX). However, when I answer it, there is absolutely no
I just did a little RTP debug and this is what it shows:
== Spawn extension (304, 301, 1) exited non-zero on 'SIP/304-d9f8'
-- Accepting AUTHENTICATED call from 216.152.244.81:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs =
That's kind of what I suspended. Either that, or in THEIR install
there was a bug and for whatever reason they can't upgrade at the
moment, which may be the case.
On 3/20/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Mar 20, 2006, at 3:47 PM, Matt wrote:
I received an e-mail from a vendor
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have a perl app that listens for hangups and then grabs the call out
of the database using the uniqueid. Maybe not the neatest way but it
works well.
Darren Wiebe
[EMAIL PROTECTED]
Mark Ackroyd wrote:
The reason for it being 0 is because as
We are faced with a problem concerning queues.
When we have several calls in different queues, is there some sort of
way to open a channel between a (sip-)phone and a SPECIFIC call in a
queue using the Asterisk manager api?
We would like to do this even when we are not a member of that
This is driving me nuts!!
After unplugging all the phones, restarting the router and the modem, and
reconfigurating my * boxes, I was finally able to communicate between both
phones only when they were both registered to the same server.
If I try to call between phones between two different
I've had terrible audio quality with IAX2 on Asterisk 1.2, there was a
thread about it some time ago and many people reported the same thing.
I think the thread was entitled 'problem with new jitterbuffer
implementation' or something similar. However, it did not appear to be
the jitterbuffer
I've had the same problem with all boxen running the same version. We
ditched IAX2 for SIP and it has been working fine since.
Doug Lytle wrote:
Barry Flanagan wrote:
Hi,
I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to
connect to a 1.2.5 box for PSTN. There are 15
Interesting. I mean I'm sure this vendor HAS an issue with IAX...
but I guess I'm just curious what brought it out and if it's an issue
with his implimentation of asterisk. At any rate.. it is odd that I
only have IAX issues to him and not someone else. Of course, I (and
the other company)
We had no problems with Asterisk-1.2 (I believe the last pre-1.2
version we ran was 1.0.8), so yes I would assume it's a problem in 1.2
only. Unfortunately we are also reliant on some of the other features
in 1.2, so we'll stick with SIP for the time being.
Matt wrote:
Interesting. I
Not sure about 1.2.4 but with 1.0.9, you'll need to add noanswer to playback since playback will try to answer the line, i.e.,exten = 5551234,1,Playback(you-wont-be-billed-for-hearing-this, noanswer)
exten = 5551234,n,Answeretc.
___
--Bandwidth and
At 07:07 AM 03/20/2006, you wrote:
I've been involved in projects with
eighty mailing lists to cover every facet of the project to try and keep
traffic down and it Simply Does Not Work. Anyone claiming otherwise has
never been involved in such a project.
I doubt the goal is to keep traffic
Hi all,
I would like to use Asterisk in the E1/R2 digital
compelled enviroment. Which card is better TE210P Dual T1/E1 card or Sangoma A102U Dual T1/E1? I heard Asterisk's
Unicall add-on can support R2 signalling. But I dont have no idea. Would you
give me advice?
Thanks for help
Ganbaa
Oops !
I have upgraded TRUNK again via SVN and all was seeming to be fine, no more
invalid IAX2 frames and able to place and receive calls.
I was happy..
But, few calls later (about 5 minutes) : INVAL frames again and no more
possibility to place or receive calls, no prompt tone, nothing !
Guys, anybody has some info regarding the format that queue_log has and how
to interpret it.. I found some info on the wiki about the conditions of a
call but the first fields I still don't know what they are for, although I
can imagine one of them is a call identifier, etc. but want to be sure.
Anton Krall wrote:
Guys, anybody has some info regarding the format that queue_log has and how
to interpret it.. I found some info on the wiki about the conditions of a
call but the first fields I still don't know what they are for, although I
can imagine one of them is a call identifier, etc.
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