Hello
Apparently, those are just warnings, but I'd like to know what those
messages mean:
[Sep 19 15:32:43] NOTICE[42559] callerid.c: Unknown IE 18
[Sep 19 15:32:43] NOTICE[42559] callerid.c: Unknown IE 21
Thank you.
___
-- Bandwidth and Colocation P
Stefan Gofferje wrote:
>[CC] chan_agent.c -> chan_agent.o
> chan_agent.c: In function ‘unload_module’:
> chan_agent.c:2496: error: void value not ignored as it ought to be
> make[1]: *** [chan_agent.o] Error 1
> make: *** [channels] Error 2
Fixed in SVN r143735.
Thanks,
--
Sean Bright
[EMAIL
Dae,
If you can assist to my session may be we can discuss this issue you
are having. I am about to add Colombia support for OpenR2, and even if
you want to stick with Unicall I'd like to see what's going on there
:-)
Guys, just for your information, as of today, there is now an
asterisk-r2 maili
Hi,
when I still had ISDN, I was using Hangup(causecode) to send e.g. "Wrong
number" to unwelcome callers.
Meanwhile, I am only using SIP providers (no PSTN lines any more) and I
would like to do similar, i.e. send specific SIP headers. Besides "wrong
number", I would especially like to send 302 t
[CC] chan_agent.c -> chan_agent.o
chan_agent.c: In function ‘unload_module’:
chan_agent.c:2496: error: void value not ignored as it ought to be
make[1]: *** [chan_agent.o] Error 1
make: *** [channels] Error 2
Terve,
Stefan
--
Last words of a stormchaser:
"Where is that rotation on the radar?
This has now happened to me on two different machines with different
hardware. Suddenly a user dials and the last port on the card will have
a loud noise and the call cannot complete.
The first machine where this happened had a TDM400P card with 4 FXO
ports and the second machine
On Friday 19 September 2008 14:54:58 George Williams wrote:
> I am issuing CLI commands via script, using the "asterisk -rx" method.
>
> Its working great. Now, I need to get the results of the command to look
> for error messages, etc.
>
> I've tried setting several "-v" flags as well, but I only
I should have added that this configuration is on a local LAN connected via
a Cisco 2900 switch.
On Fri, Sep 19, 2008 at 3:54 PM, Todd Reese <[EMAIL PROTECTED]> wrote:
> Hi All,
>
>
> I'm currently having trouble with dropped phone calls. The following error
> message is always in the log. Th
>From the "doc/sip-retransmit.txt"
What is the problem with SIP retransmits?
-
Sometimes you get messages in the console like these:
- "retrans_pkt: Hanging up call XX77yy - no reply to our critical packet."
- "retrans_pkt: Cancelling retransmi
Dear Asterisk Users,
I just wanted to remind everyone that the DruidCON early bird special ticket
pricing will be over on 20 Sept 2008, which is tomorrow. Ticket prices go up
from US$50 to US$100. Pls sign up at
http://druidcon.eventbrite.com/
Thanks
Ming
-- Forwarded message --
F
Hi All,
I'm currently having trouble with dropped phone calls. The following error
message is always in the log. This is a Grandstream GXP-2000 Firmware
1.1.6.16 . The Asterisk box is currently 1.4.22-rc5. The problem has been
occurring on other versions also.
[Sep 19 15:48:02] WARNING[1365
Hi,
I am issuing CLI commands via script, using the "asterisk -rx" method.
Its working great. Now, I need to get the results of the command to look
for error messages, etc.
I've tried setting several "-v" flags as well, but I only get the Asterisk
startup text (version, license info, etc), not
Hitesh,
Not sure if I understand your question. Let me try to explain again,
There are two thing in Asterisk,
Origination and Termination.
Origination: You have a DID (Virtual Phone line say from LA), people call
that Virtual line from anywhere in the world, and you will receive that call
to your
Hi Jai,
If I understand correctly then the DID will enable to call me on the
hardphone connected to the Asterisk. Will it also enable me to call
out using the PSTN line at my home in India from Canada?
Thanks.
Best REgards,
Hitesh
On Fri, Sep 19, 2008 at 10:33 AM, Jai Rangi <[EMAIL PROTECTED]>
Yes
I can see the channels with UC show channels, and it's says "Idle"
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales
Sent: Friday, September 19, 2008 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asteri
Hitesh,
If you dont have experience with Linux I would recommend you to use Trixbox,
that will come with all the required packages and will do everythign for
you.
Re: FXO and FXS, you don't need to buy any card for True VoIP. Now you can
buy DIDs that can come to your asterisk over the internet.
BJ Weschke wrote:
> Mark Phillips wrote:
>
>> Hi all,
>>
>> When one uses the follow-me logic to forward calls to lots of phone
>> devices do subsequent calls get routed to the VM (or whatever the 10x
>> is)?
>>
>> In other words, if I want my office, house and cell phones to ring
>> whenever a
Not really the unicall setup must be idem. So you can see the unicall
channels ?
It's moises are busy i can give you support too
Regards,
Luis Morales
On Sat, Sep 20, 2008 at 12:15 PM, Dae Yeung Um <[EMAIL PROTECTED]> wrote:
> Hi Luis,
>
> But this E1 has 30 channels, all for both direction
Hello Moises
The Telco is ETB and I'm in Colombia.
I put ar,20,4 because on all instructions found on internet says that.
Anyway, I tried co-land, with same results.
Thank you for your offer on troubleshoot remotely, but I'll attend Astricon
2008 on nextweek and I have to leave tomorrow. May be
Hello Humberto
It's cas actually...
Ccs was a transcription mistake.
Thank you
DAE
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Humberto
Figuera
Sent: Thursday, September 18, 2008 8:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
S
Hi Luis,
But this E1 has 30 channels, all for both directions...
I must differentiate this??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales
Sent: Friday, September 19, 2008 8:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussi
I'm getting this really strange permissions error when I try to start
asterisk:
# /usr/sbin/asterisk -U asterisk -G asterisk -gc
== Parsing '/etc/asterisk/asterisk.conf': Found
Running as group 'asterisk'
Running as user 'asterisk'
Asterisk 1.2.26.1 svn rev 79171, Copyright (C) 1999 -
Hello Ram,
Thanks for the response.
As I said there are too many options out there :). Could you help me
in settling down on one? Something that will work with the phone lines
in India is just fine for me.
I don't have any or much Linux experience, but willing to play around,
so any compatible d
[EMAIL PROTECTED] schrieb:
> Thanks for help, but I don't understand what you say. How is it
> possible to handle the error in the dialplan if my request return a 482
> after entering Asterisk, but before accessing the dialplan ?
Ok :). I meant, you should handle the whole thing in the dialplan
Daniel Johnson wrote:
> Hi,
>
> I have a 10 line PRI E1 ISDN service from AAPT. Connected to Asterisk
> 1.4 via a Digium TE121P.
>
> All oubound calls work fine.
>
> Inbound works only if I Dial a SIP phone directly or as the first step.
> This phone MUST NOT be busy or else the call will fail
On Fri, 19 Sep 2008, Cory Andrews wrote:
Gordon, the TE110P is disco'd. I don't believe there is any more
product in the channel on that model. Digium has replaced the TE110P
with the TE120P/TE122P(w/echo can).
OK, but I can still get the TE110P as my supplier now has stock of them,
so why
Gordon, the TE110P is disco'd. I don't believe there is any more product in
the channel on that model. Digium has replaced the TE110P with the
TE120P/TE122P(w/echo can).
Cory J. Andrews
Director New Market Initiatives
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716
On Thursday 18 September 2008 18:13:44 Jay R. Ashworth wrote:
> How those are handled if you record your own arrangement of Hey Jude once
> and loop it on music on hold, I'm not clear on.
It's exactly the same, whether it's your own recording or not, since you did
not compose the music or the lyri
The usual suspects gather in a little over an hour to chat about VoIP,
Asterisk and the price of moose meat.
Any news to announce? Are you going to Astricon? Want to hook up with
anyone there?
Join us today and let's hear about your plans. I'd like to talk about
some phones but I want to wait and
Can someone tell me in simple words what the difference between these 2
cards are, and why I should buy a TE120P (more expensive) than a TE110P.
I've bought TE120P's recently just because my supplier was out of stock
on the TE110P's, but for a single board in a server, what's the
difference?
Stefan Gofferje <[EMAIL PROTECTED]> writes:
> Interesting. I have my Asterisk with RFC-1918 IPs behid a NATting PIX
> and the FIXUP SIP of the PIX makes it very easy for me to use my * as
> server for external clients as well as as client for SIP providers.
A lot of phones come preconfigured for
Alex Balashov wrote:
> sean darcy wrote:
>> David Gibbons wrote:
>>> Sean,
>>>
>>> Try 'sip show channels' or 'sip show channel ' for the drill
>>> down. I believe the codec in use will be displayed with either command.
>>>
>>> Dave
>> Thanks that worked. Now how do I get it show the codec when I'
On 17 Sep 2008, at 23:50, Philipp Kempgen wrote:
> Tim Panton schrieb:
>> On 17 Sep 2008, at 14:57, Philipp Kempgen wrote:
>>
>>> Just a quick question
>>>
>>> ---cut---
>>> [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel
>>> of type 'IAX2' (cause 34 - Circuit/channel congest
Hi,
is it possible to use zap devices (es: old analog phones) with dundi? It
seems that a sort of zapregistration like sipregistration and
iaxregistrations to include in the extensions.conf is missing...
If yeshow?
Thank you.
Giorgio.
___
-- Ban
On Tue, Sep 16, 2008 at 3:28 PM, Guilherme Loch Waltrick Góes
<[EMAIL PROTECTED]> wrote:
> We tried to setup SIP between Asterisk and the Router, but the SIP stack in
> this IOS version is broken and causes the router to reboot.
> My biggest problem is, I can't upgrade de IOS version of the router.
Mark Phillips wrote:
> Hi all,
>
> When one uses the follow-me logic to forward calls to lots of phone
> devices do subsequent calls get routed to the VM (or whatever the 10x
> is)?
>
> In other words, if I want my office, house and cell phones to ring
> whenever a call comes in and I answer it on
Hi dae,
Your zapata.conf must be ok,
now inyour unicall.conf
[channels]
language=es
context=from-pstn
usecallerid=yes
hidecallerid=no
immediate=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=y
Igor Zamocky wrote:
> http://www.voip-info.org/wiki-Asterisk+cmd+Dial
>
> attribute "i"
>
> i: Asterisk will ignore any forwarding requests it may receive on this dial
> attempt. (new in 1.4) Useful if you are ringing a group of people and one
> person has set their phone to forwarded direct to voi
If I am dialing a phone that has had a "call forward" put on, is there
anyway to stop asterisk following the call forward ?
I have a "group" of people that are on a ringall and one of them
forwarded their phone to an extension that answered and requested a
password. Anyone calling this group obvio
On Fri, Sep 19, 2008 at 2:29 AM, Hiren Mistry
<[EMAIL PROTECTED]> wrote:
> Hear span 1 is connect with pri which is for outgoing and span 2 is connect
> with Samsung OpenServ 500 PBX for PSTN Agent.
>> Pl. any one can give me help. B'coz I have to implicitly work for
>> Outgoing call from PST
Thanks for help, but I don't understand what you say. How is it
possible to handle the error in the dialplan if my request return a
482 after entering Asterisk, but before accessing the dialplan ?
Stefan Gofferje a écrit :
Hi,
[EMAIL PROTECTED] schrieb:
I have a SIP request whi
Hi,
[EMAIL PROTECTED] schrieb:
> I have a SIP request which comes from an Asterisk and which has to
> re-enter in the same Asterisk (during the same session), but during the
> second passage in Asterisk, it send me a 482 Loop Detected. So is it a
> bug or Asterisk control the session and consid
Hi,
I have a SIP request which comes from an Asterisk and which has to
re-enter in the same Asterisk (during the same session), but during the
second passage in Asterisk, it send me a 482 Loop Detected. So is it a
bug or Asterisk control the session and considere it as a loop ? If it
is not a
Perhaps you could supply some sort of log.
On 19 Sep 2008, at 08:41, Brad wrote:
> we just did a brand new installation of asterisk 1.4 on ubuntu with
> a sagnoma t-1 card
>
> everything went smooth (other than fighting a little outbound call
> issue that we are sure is a tdm network to sagno
we just did a brand new installation of asterisk 1.4 on ubuntu with a sagnoma
t-1 card
everything went smooth (other than fighting a little outbound call issue that
we are sure is a tdm network to sagnoma issue)
inbound calls are fine
dialplan is silly basic with outbound channels set to facto
I have a problem with sending a T38 FAX over a Broadsoft server.
I am sending to a PSTN FAX so the Broadsoft server is terminating SIP
point and it should send me REINVITE for T38 but it does not. It is
just accepting the FAX transmission over G711.
My question is there some specific advertisemen
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