On 3/23/06, Michiel van Baak <[EMAIL PROTECTED]> wrote:
> > I've got Asterisk 1.2.4 running with two Junghanns QuadBRI cards using the
> > qozap driver from bristuff 0.3.0-PRE-1l. One of the cards is running in TE
> > mode, the other one in NT mode.
Hm, how weird. I'm not experiencing this.
I'm ru
Hello,
I installed Asterisk from CVS on Redhat Linux 9 and working with
chan_h323 module and g729/g723 free codecs too.
My network connection diagram:
--
X-lite/X-Pro-->Asterisk--chan_h323-->GnuGK--->AS5300-->PSTN
boldsoft*CLI> show version
Asterisk CV
Hello,
I installed Asterisk from CVS on Redhat
Linux 9 and working with chan_h323 module and g729/g723 free codecs too.
My network connection diagram:
--
X-lite/X-Pro-->Asterisk--chan_h323-->GnuGK--->AS5300-->PSTN
boldsoft*CLI> show versionAste
Oh your kidding. I have to call Answer() first? I thought Queue() implicitly
called Answer. Anyway, now that I have Answer() in there, it seems to be
behaving a lot better. Thanks for your help. I'll check it out more tomorrow.
Really... I thought Queue() and other apps called Answer() themsel
Hi,
I just updated to FC5, and used zaptel-1.2.1 as it was with my last version,
it show this error:
CC [M] /usr/local/src/zaptel/wcusb.o
/usr/local/src/zaptel/wcusb.c:1452: error: unknown field owner specified
in initializer
/usr/local/src/zaptel/wcusb.c:1452: warning: initialization from
i
Hello,
I upgraded from 1.0.9 to 1.2.5 yesterday and it went ok except
for one little change that happened in the CDRs, and it's concerning the
CallerID.
Let me explain..
With 1.0.9 I used to get this in the CDRs: "COMPANY LTDA <2153>"
Now with 1.2.5 I get one part only: "2153"
The number 2
Let's try the obvious first, how aboutexten => q_main,1,Answerexten => q_main,2,etc.Fingers crossed :)On 3/24/06, Douglas Garstang
<[EMAIL PROTECTED]> wrote:
Here you go. I'm not sure this is much use. It's a bit hard to explain as I have one system calling another via IAX where the Queue() comman
I think your requirements are different to ours. I need each agents phone to
ring for a given period of time, once. Once each agent has been tried once, the
caller should drop out of the queue. I can't set the queue timeout to 0... we
don't want people to be stuck in the queue forever, or even
Here you go. I'm not sure this is much use. It's a bit hard to explain as I
have one system calling another via IAX where the Queue() command is
executed... Calls are VOIP->VOIP, on our network... This case below is where
each agent (there's 6) is rung for 30sec, but the Queue aborts after 120s,
No, I used the term stupid because it was easy to type and it's been a
long day :)
Aaron
On Thu, 23 Mar 2006, Douglas Garstang wrote:
If you referring to me Aaron, I don't recollect every using the word 'stupid'.
I have said many times that the way certain things are done for an Enterprise
I'm not looking at this project. It's already been chosen.
From this perspective, maybe you should give a heads up that hey, this
project isn't worth the time and effort that they're having you put into
it.
Have you ever considered that maybe this is because there's plenty of things
not to l
On Mar 23, 2006, at 6:58 PM, BJ Weschke wrote:
We run into situations like this often as well, and it's truly
unfortunate, because it gives our industry and the technology driving
it a bad name, and like you, some customers want to go back to TDM and
have nothing to do with VoIP at all because
Queues don't process the dial-plan, period. When you pass a call to a
queue, it's either going to be answered by an available agent, or stay
in the queue until the queue time-out has been reached. As you have
already seen, the agent time-out value will automatically log out an
agent who doesn
In that case, you should post part of extensions.conf which is related to Queues/agents, which makes it easier to troubleshoot your problem. Besides, you haven't mentioned how incoming calls get into your queue. PSTN or VoIP calls?
All providers set some limit on the time a call can be placed on th
So your Polycom 501's will eventually re-subscribe and BLF will eventually
start working again after a reboot using your patch? How long will that
take? Is the time to re-subscribe something you can set on the phone?
That would be quite acceptable to me if the phone eventually re-subscribed
on i
On Mar 23, 2006, at 11:05 AM, Ronald Lewis wrote:
After months of BroadVoice ignoring my trouble tickets for dropped
calls, delayed termination, etc., I'm throwing in the towel. While
they have credited $19.95 to my account, they refuse to credit
anything more, despite ALL of the problems I'v
If you referring to me Aaron, I don't recollect every using the word 'stupid'.
I have said many times that the way certain things are done for an Enterprise
Class piece of software are unacceptable, or maybe reacted in incredulation
that it would even be considered being done in a certain way.
>
>*SCREAMS*
>
>Tell me, why is someone as sarcastic as you and with such a caustic
>attitude towards an OPEN SOURCE project that is maintained and fixed
>primarily by people on their OWN TIME even looking at this project.
I'm not looking at this project. It's already been chosen.
>
>Day in and
I don't see why I'd be getting CHANUNAVAIL. As I said, all the agents are
logged in and are in astdb and 'sip show peers'.
I just had another go, I passed a timeout of 300 to the Queue() command and set
timeout=30 in queues.conf for the queue. The Queue timed out after TWO minutes
(120s) and o
That's not different. I made a mistake stating it was sip reload instead of
just reload.
> -Original Message-
> From: Douglas Garstang [mailto:[EMAIL PROTECTED]
> Sent: Thursday, March 23, 2006 5:29 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asteris
Yipes, Aaron. The point of a troll is to elicit exactly the behavior you
just exhibited.
If everyone were to adopt a really simple protocol: "Ask nicely, Doug, or
we'll all remain silent," he would either start behaving or go away.
I'm sure all the tumult he elicits is a vast ego boost for h
Hi Antonio,
Thanks for the reply and for the links.
Regards,
Ryan
At 09:14 PM 3/23/2006, Antonio Rabena wrote:
You can try using asterisk-addons
http://www.voip-info.org/wiki/view/Asterisk+voicemail+database or
asterisk realtime
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voi
"300" is the maximum time a call can be placed in a queue, after that, next priority will be executed."15" is the time each agent will be rung before * tries the next agent."CHANUNAVAIL" happens since Queue rings Agent/xyz, Agent/xyz in turn tries to ring SIP/abc but cannot find it, hence...
BTW, y
I have more than 100 users with out problemand i'm using the file no db.On 3/23/06, Antonio Rabena <
[EMAIL PROTECTED]> wrote:You can try using asterisk-addons
http://www.voip-info.org/wiki/view/Asterisk+voicemail+database orasterisk realtimehttp://www.voip-info.org/wiki/view/Asterisk+RealTime+Voic
Aaron Daniel wrote:
*SCREAMS*
Tell me, why is someone as sarcastic as you and with such a caustic
attitude towards an OPEN SOURCE project that is maintained and fixed
primarily by people on their OWN TIME even looking at this project.
Yipes, Aaron. The point of a troll is to elicit exact
I have here de backtrace resultUsing host libthread_db library "/lib/libthread_db.so.1".Core was generated by `asterisk -g'.Program terminated with signal 11, Segmentation fault.
#0 0xb7ece142 in ?? ()As I see it was in the libthread library.. So can it confirmmy theory that is a
Thanks, but I don't really follow your reply. Agents are being logged in with
AgentCallBacklogin. If I do a 'show agents' command, I see them... all nice and
logged into their queues...
-Original Message-
From: CC Jay [mailto:[EMAIL PROTECTED]
Sent: Thu 3/23/200
You can try using asterisk-addons
http://www.voip-info.org/wiki/view/Asterisk+voicemail+database or
asterisk realtime
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail
At 04:51 AM 3/23/2006, you wrote:
Hi,
Is there a howto to do this? I'm using voicemail.conf and
sip.con
I have here de backtrace resultUsing host libthread_db library "/lib/libthread_db.so.1".Core was generated by `asterisk -g'.Program terminated with signal 11, Segmentation fault.
#0 0xb7ece142 in ?? ()As I see it was in the libthread library.. So can it confirmmy theory that is a m
Rather than continue to go in circles with queues, I'll ask upfront.
What exactly does the timeout option in queues.conf specify? The docs
contradict themselves on the meaning. What exactly does the timeout option
passed to Queue()? Docs are flaky there also.
I have a BIZARRE situation where I'
On 3/23/06, Alvaro Parres <[EMAIL PROTECTED]> wrote:
> Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAM
> about 75 Polycom Phones, one E1 for incoming calls.
>
> We have program a page system with the page command and the auto answer
> funtion
> of polycom.
>
> We have detect v
Hi,
Is there a howto to do this? I'm using voicemail.conf and
sip.conf for my voicemail users. Does it really has a limit?
Thanks,
Ryan
At 08:23 PM 3/23/2006, Antonio Rabena wrote:
How about moving your voicemail users into db?
At 03:50 AM 3/23/2006, you wrote:
Hi,
Is there
A queue timeout of 300 isn't really what I'm trying to do.
I want to ring each agent once, for a specified period of time, say 15s, and
then exit the queue. I've tried setting timeout=15 (which the docs are
contracting on the meaning off), and set my queue timeout to agents in queue x
15 n
We ran a system at one site with 2 TDM400's in it to hook up to 8 analog mobile
phone gateways.
Asterisk was much more reliable than the analog phone gateways, but we still
rebooted it once a week.
Running on a dual athlon 1800 we picked up very cheaply.
regards,
Paul Hales
Technical Manager
You still need to bind Agent/* with SIP/* via AgentLogin/CallbackLogin/etc. to make it work (i.e., solving the CHANUNAVAIL prob.)
then the timeout DOES appear to be honored, except that Queue returns CHANUNAVAIL, and does not proceed to the next step in the dialplan. Again, Why?
I'm just trying to
The magic command you want is 'WaitExten' - 'show application waitexten'
on the asterisk command line. 'show applications' is also a good one.
--Rob
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dinesh Nair
> Sent: Sunday, 19 Marc
You may wanna try...extensions.conf:exten => q_main,1,Queue(oneeighty_main300)...[oneeighty_main]musiconhold = defaulttimeout = 15...Cheers.
On 3/24/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
Egads. Getting queues to work is like pulling teeth.extensions.conf:exten => q_main,1,Queue(oneeig
Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAMabout 75 Polycom Phones, one E1 for incoming calls.We have program a page system with the page command and the auto answer funtionof polycom.
We have detect via diaplan if the phone isn't in call we place the call. All this via Ma
I would like to hear from anyone good or bad as what their experience has
been in recent times with STABILITY of current builds of Asterisk and
drivers for TDM400P.
The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P cards.
I am not concerned with: price points, or the advantages
How about moving your voicemail users into db?
At 03:50 AM 3/23/2006, you wrote:
Hi,
Is there an account limit for voicemail? I have 80+ users
in the voicemail and I can only reach the 70-ieth user. If there is
a limit how can I increase it to hundred for example?
Thanks,
Ryan
___
On Mar 23 2006, Steve Gladden wrote:
Sorry for being a bit of a newbie here but I find the
docs or README for downloading the G.729 codec from Digium
are not as detailed as I would like or just don't really
break down the different versions to a point that I am clear
on which one to grab.
The c
rhino channel bankOn 3/23/06, C F <[EMAIL PROTECTED]> wrote:
Carrier Access Adit 600On 3/23/06, Curt Shaffer <[EMAIL PROTECTED]> wrote: Is anyone out there using FXS channel banks to connect analog phones to
> Asterisk? If so do you have brand recommendations?>> Thanks Curt> ___
Carrier Access Adit 600
On 3/23/06, Curt Shaffer <[EMAIL PROTECTED]> wrote:
>
>
>
> Is anyone out there using FXS channel banks to connect analog phones to
> Asterisk? If so do you have brand recommendations?
>
>
>
>
>
> Thanks
>
>
>
> Curt
> ___
> --Ban
> Does anyone have the polycom soundpoint ip's successfully remotely
> provisioning? I've got the phone pulling default configs, and it's
> downloading phone specific information, but it's not actually using
that
> information. Any help would be appreciated :)
>
> --
> Aaron Daniel
Aaron,
I
Hi,
I am working on a H.323 project which involves GnuGk and Asterisk My
current goal is to provide IVR functionality for the H.323 users which
register through GnuGk(eg. call credit information)
I have successfully built a H.323 platform using GnuGk - it uses SQL
accounting and authorisatio
Hi,
Is there an account limit for voicemail? I have 80+ users in the
voicemail and I can only reach the 70-ieth user. If there is a limit
how can I increase it to hundred for example?
Thanks,
Ryan
___
--Bandwidth and Colocation provided by Easynews
notransfer=yes
It prevents Asterisk of getting out the media-path.
-Original Message-
From: Matt <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Cc:
Sent: Thu, 23 Mar 2006 21:57:33 -0500
Delivered: Thu, 23 Mar 2006 21:12:08
Subject:[Aste
It appears that when realtime is enabled in queues.conf
persistantmembers no longer has effect on dynamically added members. I am
wondering if this is a intended or a bug.
___
--Bandwidth and Colocation provided by Easynews.com --
Aster
Aaron Daniel wrote:
Yeah, everything but the individual phone configuration is working fine.
Make sure the .cfg files are pointing to a custom
phoneX.cfg file, not the template. That got me in the beginning. *blush*
--
National Manager - Special Projects
< Melbourne / Sydney / Canberra / Ho
Yeah, everything but the individual phone configuration is working fine.
The logs are uploading and everything. I'll look at those more closely
tomorrow.
Aaron
On Fri, 24 Mar 2006, Avi Miller wrote:
Aaron Daniel wrote:
voicemail stuff is working, just not the registration information, unles
Do you happen to now a modell number or a link for this phone?
Thank,
Kevin
On 3/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
> On Fri, 2006-03-24 at 13:07 +1100, James Harper wrote:
>
> > I haven't done any sort of research, but I've been told that GSM+DECT
> > phones are available, and while h
On 3/23/06, Rick Smith <[EMAIL PROTECTED]> wrote:
> Customer got ripped off by a previous VOIP provider and had a REAL
> distaste for VOIP, even done right...
>
> Get this
>
> they had a SIP Server in San Diego, with 25 phones in NYC and another 20
> in Atlanta.
>
> Average hops were 24, and ov
I have a user who is off my system with IAX. When he calls and goes
out my long distance provider my asterisk switch seems to be bridging
the two calls. As a result I loose all accounting information. All I
get is the call setup time (15 or 20 seconds).
How can I either make asterisk not bridge
AH HA! The firmware was sadly out of date... I upgraded the firmware
and the interface looks like a different phone!
On 3/23/06, mustardman29 <[EMAIL PROTECTED]> wrote:
> Do you have the latest firmware?
>
> Are you trying to use the phone with a hunt group or do you want each line
> to work wit
Are these MGCP or SIP 501 phones?
On 3/23/06, Rick Smith <[EMAIL PROTECTED]> wrote:
>
> Converted a strictly VOIP system in NYC to NEC IPK TDM system...
> will have 25 Polycom 501's for sale.
>
> Best offer, offlist only please.
>
> R
> ___
> --Bandwidth
if you want to use the defaults unavailable and bussy just put it in /var/spool/asterisk/voicemail/default/100/unavail.wav
The "100" extension will automatically be created after you leave your
first voicemail message, change "unavail.wav" for "busy.wav" to use
them as
Voicemail(u100) and
Aaron Daniel wrote:
voicemail stuff is working, just not the registration information,
unless I missed something.
Do you have individual .cfg files in the root of the FTP
server for each phone, pointing to custom phoneX.cfg files with specific
registration info for that phone? You should see
On Fri, 2006-03-24 at 13:07 +1100, James Harper wrote:
> I haven't done any sort of research, but I've been told that GSM+DECT
> phones are available, and while having them seamlessly switch network
> types during a call probably isn't possible, they can function as a
> cordless handset.
>
> Can
On 3/23/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> I don't know why the situation is different, but we've been using Polycom
> phones with BLF, and it's ok. I'm using Asterisk 1.2.5, and a 'reload' will
> clear sip subscriptions and BLF, but a 'sip reload' does not.
>
It's different beca
Thanks for the insight.. I have been considering broadvoice because it seems
to be a "big" name, but yours is not the first negative feedback.. Perhaps
I'll stay w/ Vonage for the time being - I hate having it go analog and back to
digitial in the space of 5 feet of cable, but at least it work
Customer got ripped off by a previous VOIP provider and had a REAL
distaste for VOIP, even done right...
Get this
they had a SIP Server in San Diego, with 25 phones in NYC and another 20
in Atlanta.
Average hops were 24, and over 210 ms end to end.
Just poor engineering, and they didn't
Hadley Rich wrote:
On Friday 24 March 2006 12:53, Larry Alkoff wrote:
What do I have to do to dial an exten -> with the dial command in it?
Asterisk isn't recognizing commands in my newly created [context].
There is a really good book available here[1] that will answer this and a lot
of other
Yes I reload each time.
Larry
Aaron Daniel wrote:
Did you reload the dialplan in the CLI? I think it's extensions reload.
That'll refresh your settings... If that doesn't work, post your
dialplan so we can see what's going on :)
Aaron
On Thu, 23 Mar 2006, Larry Alkoff wrote:
That's how I _
Is anyone out there using FXS channel banks to connect
analog phones to Asterisk? If so do you have brand recommendations?
Thanks
Curt
smime.p7s
Description: S/MIME cryptographic signature
___
--Bandwidth and Colocation provided
*SCREAMS*
Tell me, why is someone as sarcastic as you and with such a caustic
attitude towards an OPEN SOURCE project that is maintained and fixed
primarily by people on their OWN TIME even looking at this project.
Day in and day out, you find something you don't like, and spend a week
bitc
Larry Alkoff wrote:
What do I have to do to dial an exten -> with the dial command in it?
Asterisk isn't recognizing commands in my newly created [context].
You must send the call into the context. That can be done in the
following ways:
a. Device context - e.g. in your sip.conf. You have so
And also curious why a K6-3? but not a K6-2?
And then of course why do we get a K6-3 but not an Athlon?
:-)
Steve
> Sorry for being a bit of a newbie here but I find the
> docs or README for downloading the G.729 codec from Digium
> are not as detailed as I would like or just don't really
> break
> Steve,
>
> Excellent explanation.
>
> In a nutshell, it might be better to just use a phone that can
> automatically switch between GSM and WiFi. Of course, that's limited
to
> handful of handsets.
I haven't done any sort of research, but I've been told that GSM+DECT
phones are available, and
Aaron Daniel wrote:
Does anyone have the polycom soundpoint ip's successfully remotely
provisioning?
Yup. I have two sites with 5 phones at each site successfully
configuring themselves off my central FTP server. The next site (50
phones) is being installed next week. :)
--
National Mana
The file downloads are showing up in the log files, at least from what I
can tell. The phone's pulling specific information, and I know the
voicemail stuff is working, just not the registration information, unless
I missed something.
Aaron
On Thu, 23 Mar 2006, C F wrote:
How do you know it
Why the changeover back to TDM??
- Gabe
- Original Message -
From: "Rick Smith" <[EMAIL PROTECTED]>
To:
Sent: Thursday, March 23, 2006 4:39 PM
Subject: [Asterisk-Users] Polycom 501's for sale
>
> Converted a strictly VOIP system in NYC to NEC IPK TDM system...
> will have 25 Polycom
On Friday 24 March 2006 12:53, Larry Alkoff wrote:
> What do I have to do to dial an exten -> with the dial command in it?
> Asterisk isn't recognizing commands in my newly created [context].
There is a really good book available here[1] that will answer this and a lot
of other questions easily a
On 3/23/06, mustardman29 <[EMAIL PROTECTED]> wrote:
> Thanks BJ,
>
> I tried your patch and it worked fine for me so thank you so much for the
> effort. It is very much appreciated. Especially since I am not capable of
> coding myself.
>
> Unless I can get a total solution so that it just works n
I don't know why the situation is different, but we've been using Polycom
phones with BLF, and it's ok. I'm using Asterisk 1.2.5, and a 'reload' will
clear sip subscriptions and BLF, but a 'sip reload' does not.
Doug.
> -Original Message-
> From: mustardman29 [mailto:[EMAIL PROTECTED]
Olle E. Johansson said:
> Be careful with reloads out there. They do affect your PBX and
> they're meant to.
Does every type of reload instruction causes the subscriptions to be lost?
Or is it a specific type of reload that causes it?
Will "extensions reload" or "reload chan_sip.so" for examp
Polycom's can work in one of two ways:
a) using self configuration
b) downloading it from a ftp server
To make your Polycoms work with Asterisk you actually don't need the
phone to download any configuration, with the one embeded is ok. In any
case, when turned on, the phone searches for t
This is something that the phone you are using will have to support,
and your case with Cisco phones it is NOT supported, the Polycoms
support this. If you use a something like FOP (Flash Operators Panel)
then you could define an extension that just goes to VM, and drag that
call to VM.
On 3/23/0
How do you know it's pulling those files? Is there an error reported
by the phone reading those files? Maybe a typo in the xml files?
On 3/23/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:
> Does anyone have the polycom soundpoint ip's successfully remotely
> provisioning? I've got the phone pulling
Sorry for being a bit of a newbie here but I find the
docs or README for downloading the G.729 codec from Digium
are not as detailed as I would like or just don't really
break down the different versions to a point that I am clear
on which one to grab.
The choices for 32bit are:
drwxr-xr-x3 0
Did you reload the dialplan in the CLI? I think it's extensions reload.
That'll refresh your settings... If that doesn't work, post your dialplan
so we can see what's going on :)
Aaron
On Thu, 23 Mar 2006, Larry Alkoff wrote:
That's how I _thought_ it worked but extens in such a created [con
Think a zaptel recompile is just what you need.Alyed
Return-Path: <[EMAIL PROTECTED]> Thu Mar 23 17:05:27 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Thu, 23 Mar 2006 17:05:27 -0700
i've got a aste
Egads. Getting queues to work is like pulling teeth.
extensions.conf:
exten => q_main,1,Queue(oneeighty_main1)
exten => 80014055,1,Dial(SIP/80014018,15,tr)
exten => 80014057,1,Dial(SIP/80014018,15,tr)
exten => 80014052,1,Dial(SIP/80014018,15,tr)
queues.conf:
[oneeighty_main]
musiconhold = def
If you want stable use external gateways and two servers
set up to fail over to each other.
Second best is two two T1 cards so you have something
left when one of them fails.
If you want to bet your job on a 4 T1 card, Sangoma
has excellent echo cancellation and a million hour MTBF.
Pavel / Joseph - I think I have something that will solve your dilemma.
Powersense make an IEEE 802.3af to Cisco Discovery Protocol (CDP) converter
which should work with your Netgear switch. You need (1) for each Cisco
phone. You run RJ45 out of your Netgear PoE Switch, into the PowerSense
m
What do I have to do to dial an exten -> with the dial command in it?
Asterisk isn't recognizing commands in my newly created [context].
Larry
C F wrote:
Yes, you just create it.
On 3/23/06, Larry Alkoff <[EMAIL PROTECTED]> wrote:
It _appears_ that the only way to create valid [context] is b
On 3/23/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:
> Does anyone have the polycom soundpoint ip's successfully remotely
> provisioning? I've got the phone pulling default configs, and it's
> downloading phone specific information, but it's not actually using that
> information. Any help would be
Converted a strictly VOIP system in NYC to NEC IPK TDM system...
will have 25 Polycom 501's for sale.
Best offer, offlist only please.
R
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update o
On Thursday 23 March 2006 14:54, Douglas Garstang wrote:
> Ok Andrew. Here's one for you... I just changed qualify from yes to no in
> the database... a 'sip show peers' still showed Asterisk as qualifying the
> users... I had to do a reload to get to accept the change to the database.
Aaron's alr
> I have several locations, each connected by a Sonicwall VPN through
> PPPOE DSL, with Snom 360 phones.
>
> I've found that I have to tweak the Asterisk server MTU (inside one of
> the firewalls) to get everything to work "just right". Set the server
> MTU too low, and the Snom phones don't commu
Thanks BJ,
I tried your patch and it worked fine for me so thank you so much for the
effort. It is very much appreciated. Especially since I am not capable of
coding myself.
Unless I can get a total solution so that it just works no matter if I
reload or reboot then it's not really a solution
Please don't tell me what I think you are. Are you saying that to change a
configuration setting for the phone I have to remove it as a peer, and then
wait for it to re-register? Are you serious???
Doug.
> -Original Message-
> From: Aaron Daniel [mailto:[EMAIL PROTECTED]
> Sent: Thursda
Hi,
which OSX softphone do you use that supports IAX2 protocol with Asterisk?
thanks
Mike
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I've tried adding [new_context] with extens defined in extensions.conf
but they are not recognized or dialed out.
Are you saying I can add various
context=new_context1
context=new_context2
etc
lines in sip.conf [general] and that's all there is to it?
I'm not sure what you mean about "per device
Dr. Michael J. Chudobiak wrote:
Hi all,
I have several locations, each connected by a Sonicwall VPN through
PPPOE DSL, with Snom 360 phones.
I've found that I have to tweak the Asterisk server MTU (inside one of
the firewalls) to get everything to work "just right". Set the server
MTU too
Do you have the latest firmware?
Are you trying to use the phone with a hunt group or do you want each line
to work with a different TISP? I think if you want all 3 lines to work on a
hunt group you only configure line one as global. It's quite well
documented but I haven't messed around with
That's how I _thought_ it worked but extens in such a created
[context_name] are not seen or used by Asterisk to dial out.
There is something missing.
Larry
Aaron Daniel wrote:
Yes.
Just create a context that you want the phones to dial from in
extensions.conf.
[context_name]
exten => 123
BJ Weschke wrote:
On 3/23/06, Larry Alkoff <[EMAIL PROTECTED]> wrote:
It _appears_ that the only way to create valid [context] is by a
context = line in sip.conf.
Is there another way to create a [new_context] in extensions.conf so I
can dial from it?
Right now most of my extens are in [defaul
Steve,
Excellent explanation.
In a nutshell, it might be better to just use a phone that can
automatically switch between GSM and WiFi. Of course, that's limited to
handful of handsets.
Steve Kennedy wrote:
On Thu, Mar 23, 2006 at 01:48:16PM +0100, Tomislav Parina wrote:
In article <
Thanks Olle,
So am I to understand that you are under the impression that BLF DOES work
after a reload or reboot provided the phone re-registers?
I have two separate manufacturers phones. Aastra 9133i and Grandstream
GXP2000 and both behave EXACTLY the same way. After a reload or reboot, BLF
st
Luigi Rizzo wrote:
On Thu, Mar 23, 2006 at 01:18:15PM -0600, Larry Alkoff wrote:
It _appears_ that the only way to create valid [context] is by a
context = line in sip.conf.
Is there another way to create a [new_context] in extensions.conf so I
can dial from it?
manually with an editor ?
or
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