What is a UA? I am not using an ATA-186 or a Cisco 7960. The only
Asterisk related hardware that I am using is TDM 400P and X100P.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Wednesday, June 04, 2003 5:27 PM
To: [EMAIL PROTECTED]
Please I need help, I don't know why,almost every time I dial on my dect
phones, the dialtone doesn't go off and * doesn't recognise anything I'm
using two dlink voip gateways, MGCP: DL102s. Any ideas?
thanks in advance
michelle matis
-
Tu cuenta de correo gratuita Mixmail
this might be a better dump:
#0 0x41ec7279 in ast_oh323_new (i=0x810e538, state=0) at chan_oh323.c:1170
#1 0x41ec786e in oh323_request (type=0x4971aa6c OH323, format=8,
data=0x4971a91c) at chan_oh323.c:1302
#2 0x0805878f in ast_request (type=0x4971aa6c OH323, format=4,
data=0x810e538)
michelle matis litio wrote:
Please I need help, I don't know why,almost every time I dial on my dect
phones, the dialtone doesn't go off and * doesn't recognise anything I'm
using two dlink voip gateways, MGCP: DL102s. Any ideas?
thanks in advance
michelle matis
-
Tu cuenta de correo
Dave Alan Caruana wrote:
this might be a better dump:
#0 0x41ec7279 in ast_oh323_new (i=0x810e538, state=0) at chan_oh323.c:1170
#1 0x41ec786e in oh323_request (type=0x4971aa6c OH323, format=8,
data=0x4971a91c) at chan_oh323.c:1302
#2 0x0805878f in ast_request (type=0x4971aa6c OH323,
On Thu, 05 Jun 2003 11:25:19 +0300
Michael Manousos [EMAIL PROTECTED] wrote:
Eduardo Goncalves wrote:
Hi,
I have an ansterisk and a cisco 827-4v registered to a Gatekeeper.
asterisk has two extensions:
exten = 223,1,Dial,OH323/[EMAIL PROTECTED]
Is 223 a registered
Do you really have the channels in asterisk ?
zap show channels
Is the alarm on the E1 circuit ?
Martin
On Thu, 5 Jun 2003, Jay Banda wrote:
Hello All.
Does anyone have experience with the Valiant Comms vcl30 channel
and the Digium E100P in asterisk ? We have the vcl30 channel bank,
Eduardo Goncalves wrote:
On Thu, 05 Jun 2003 11:25:19 +0300
Michael Manousos [EMAIL PROTECTED] wrote:
Eduardo Goncalves wrote:
Hi,
I have an ansterisk and a cisco 827-4v registered to a Gatekeeper.
asterisk has two extensions:
exten = 223,1,Dial,OH323/[EMAIL PROTECTED]
Is 223 a
Asterisk's standard queue agent application requires that the phone stay off hook the whole time.
You get a beep in your ear when a call comes in, not a ring.
Bill
Reece Anderson wrote:
Hi,
We recently installed asterisk too replace our office PABX, however we
are finding it hard to get
Also, are all the fxo channels connected to the t1 channels on the channel
bank? I believe you have to map them just like on the CAC Adits...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Thursday, June 05, 2003 10:40 AM
To: [EMAIL
I had a very recent version of asterisk,
but to be sure just downloaded the latest from CVS
and compiled all packages except OH323 which
is about 3 days old ...
thanks
Dave
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 05, 2003
On Tue, 2003-06-03 at 13:53, Steve Underwood wrote:
Q.SIG is a PBX to PBX variant to ISDN. If you can change the config of
your switch you can probably make it do CTR4 (EuroISDN, Net5, and
various other names), which the E400P does support. Then life will
become much easier.
Hopefully not
http://www.dynx.net/ASTERISK/DOCS/RTF/agentlogin.rtf
Reece Anderson wrote:
Hi,
We recently installed asterisk too replace our office PABX, however we
are finding it hard to get documentation on the way Agents login.
In the agents.conf we have setup a user of agent = 1003,,Test. In
I've posted 2 shots of the CID spectrum on
http://vmail.espia.it/shots/ , for those
that want to dig more in the issue.
Matteo.
Il gio, 2003-06-05 alle 16:12, Brancaleoni Matteo ha scritto:
Hi 2 all.
I've a nice issue my * system.
I have:
asterisk latest cvs
T100P + Zhone cb (3fxo+10fxs
The updated Budgetone firmware (1.0.3.60) has indeed fixed the silent
DTMF issue.
By the way, Grandstream just got the silent DTMF problem fixed for me
and sent me an updated revision this morning (1.0.3.60). I am just
about to install it, but it may require that I debug my tftp server,
which I
Hi,
I have an incoming call that I would like answered every
time by a different SIP phone (out of 50).
Also, some of the phone may not be available (may be turned
off and thus unregistered with Asterisk).
Any way of doing this?
Paulo H. Mannheimer
On Thu, 5 Jun 2003, Paulo Mannheimer wrote:
I have an incoming call that I would like answered every time by a
different SIP phone (out of 50).
hmm... pass the call through an AGI first, that picks a random number
and then pass the call to that SIP phone number
Also, some of the phone may
Sorry for my terminology assumptions.
UA = User Agent, which is what the ATA-186 and 7960 are. Anything
that normally is what the end user has on their desk or on their
computer (in the case of a softphone) is considered a UA.
So, since you have both an ATA-186 and Cisco 7960, make the
chan_h323 doesn't have this problem, I have many systems using G.729.
Jeremy McNamara
Eduardo Goncalves wrote:
Is 223 a registered alias of 827?
Have you configured this destination pattern in 827?
Yes and yes :~
I can call from the others cisco to this 827. I can't call from
On Thu, 05 Jun 2003 18:08:50 +0300
Michael Manousos [EMAIL PROTECTED] wrote:
That's because you are using G.729. Don't!
Try the same with G.711.
Thanks Michael, using G.711 asterisk and cisco worked well...
But I have to use a low bit rate codec, for use with slow links.
The olny way is
I used to have email notification working with my voicemail services but
it stopped working when I installed the new version of asterisk.
I have not changed my voicemail.conf file, so I'm out of ideas.
Does asterisk use Sendmail to send messages, or does it have its own
method for sending email?
I updated the version of asterisk I was using and the problem seems to
have been solved.
Thanks for the help
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Thursday, June 05, 2003 3:31 PM
To: [EMAIL PROTECTED]
Subject: RE:
It does use sendmail. Which app are you using ?
voicemail or voicemail2 ?
Martin
On Thu, 5 Jun 2003, Derek Beaumont wrote:
I used to have email notification working with my voicemail services but
it stopped working when I installed the new version of asterisk.
I have not changed my
It uses sendmail. Try sending a test message from the computer first,
to rule out asterisk as a problem.
--Mike
On Thursday, June 5, 2003, at 01:05 PM, Derek Beaumont wrote:
I used to have email notification working with my voicemail services
but
it stopped working when I installed the new
Simon J Mudd wrote:
Using the LD_LIBRARY_PATH as you explain appears to be rather a hack.
LD_LIBRARY_PATH is a standard Linux environment value, so complain to Linus.
I wasn't sure WHY you frown so heavily on the vendor/distribution
pwlib/h323 libraries (do they change that much or are
Dare I ask?
I'm new to Asterisk. I like what it has to offer,
however, I'm having a hard time finding documentation to configure it correctly.
Can anyone tell me where I can get good Asterisk
documents? Here's what I have put together:
http://www.simplifiednetwork.com/asterisk.
Seng
If you switched to qmail make sure you copy the sendmail script to a bin directory
Mike Reiling wrote:
It uses sendmail. Try sending a test message from the computer first,
to rule out asterisk as a problem.
--Mike
On Thursday, June 5, 2003, at 01:05 PM, Derek Beaumont wrote:
I used to have
Ok.
enabling zaptel modules debug, I found that when Eicon Diva
receives a call the T100P slips a lot.
Idem with another (newer) T100P.
The diva generates 100 irq (more or less) per second, only
when on a call.
Here's the kernel log :
Jun 6 00:19:38 asterisk kernel: New offset: 28
Jun 6
Ok. It's definately not asterisk's problem.
The problem is with the aliases used with sendmail.
The error I see is:
Cannot rebuild aliases: no database format defined
Cannot create database for alias file /etc/mail/aliases
I guess it's off to the sendmail forums
Thanks for the help
Thanks, very good insights.
The proposed method has a single flaw - it's very difficult to detect
that all SIP channels are busy, and thus queue the call.
It's a petty that SIP does not support call groups, it would make it
automatic.
Best,
PHM
-Original Message-
From: [EMAIL
quote who=Jeremy McNamara
Simon J Mudd wrote:
Using the LD_LIBRARY_PATH as you explain appears to be rather a hack.
LD_LIBRARY_PATH is a standard Linux environment value, so complain to
Linus.
This is the environment variable used to loading share libraries from
non-standard locations. This
Hello all,
While I am doing the two SIP phones call via *, I got following message:
WARNING[6151]: File chan_sip.c, Line 409 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 1 (Response)
While I am doing two H323 phones call via *, I got following message:
I was trying to be funny, guess I failed :(
Jeremy
Robert Hajime Lanning wrote:
quote who=Jeremy McNamara
Simon J Mudd wrote:
Using the LD_LIBRARY_PATH as you explain appears to be rather a hack.
LD_LIBRARY_PATH is a standard Linux environment value, so complain to
Linus.
I haven't tried this SIP features, but in the latest sip.conf.sample,
this is included:
;[pingtel]
;type=friend
;username=pingtel
;secret=blah
;host=dynamic
;qualify=1000 ; Consider it down if it's 1 second to reply
;callgroup=1,3-4
;pickupgroup=1,3-4
;defaultip=192.168.0.60
I'm uncertain if it must be a Skinny upgrade. I have upgraded
several _ancient_ Cisco 79xx phones, and I've had success by going to
SIP image of P0S30203.bin first, and then going to a more recent
image (P0S30440.bin, as an example.)
So, while upgrading with a bootstrap of a Skinny image may
[EMAIL PROTECTED] (Jeremy McNamara) writes:
Simon J Mudd wrote:
Using the LD_LIBRARY_PATH as you explain appears to be rather a hack.
LD_LIBRARY_PATH is a standard Linux environment value, so complain to Linus.
I'm aware of what LD_LIBRARY_PATH is for (and that is works on various
Eduardo Goncalves wrote:
On Thu, 05 Jun 2003 18:08:50 +0300
Michael Manousos [EMAIL PROTECTED] wrote:
That's because you are using G.729. Don't!
Try the same with G.711.
Thanks Michael, using G.711 asterisk and cisco worked well...
But I have to use a low bit rate codec, for use with slow
i've been having a problem getting two SIP phones
to bridge running through asterisk, actually one is
a SIP softphone, SJ Phone, and the other is the
Go2Call calling gateway.
Someone suggested that I don't have the right codecs.
How do I find out which codecs are installed, and how
can I install
If you have the package available for download for free from SJLabs, then
you only have G.711 codec installed on SJPhone.
If you are a developer, you can register for a G.729 codec from SJLabs.
BR,
Dan
P.S. Have you tried X-Lite? It has G.711ulaw, G.711a law, GSM and iLbc.
- Original
Hi Martin
There are no alarms on the E1 circuit, and see below the
output of zap show channels
=
*CLI zap show channels
Chan. Num. Extension ContextLanguage MusicOnH
1trunklocal en default
2
Hi Don't know if someone answered this yet...
when calling the dialapp append the |t to it
exten=sip,1,Dial(SIP/sipphone)|t
This will allow you to hit # and then the callparking extension.
Thanks
Dave
On Tue, 2003-06-03 at 09:10, denon wrote:
Hi all,
I've got a fairly minor question, but
Tielman,
You can take a look at the quick and dirty guide I'm slowly putting together if you
like...
http://www.automated.it/guidetoasterisk.htm
I'd appreciate any feedback you have on it.. and if it helped
Andy
*** REPLY SEPARATOR ***
On 06/06/2003 at 14:17 Tielman
[EMAIL PROTECTED] (Dan) writes:
P.S. Have you tried X-Lite? It has G.711ulaw, G.711a law, GSM and iLbc.
From looking again at the docs on digium's site it seems that NM
wasn't a good choice of softphone for configuring with *. basically
due to the codecs available.
I've downloaded both X-Lite
i've installed X-lite, can't get it to actually dial a SIP number,
seems cryptic compared to SJPhone ..
I have a feeling my problems is the codecs within *
though, my question was how could I know which codecs
* supports, and how to add other ones ..
cheers
Dave
- Original Message -
It should be exten=sip,1,Dial(SIP/sipphone)||t
Martin
On 6 Jun 2003, Dave Wolven wrote:
Hi Don't know if someone answered this yet...
when calling the dialapp append the |t to it
exten=sip,1,Dial(SIP/sipphone)|t
This will allow you to hit # and then the callparking extension.
Thanks
I added the line allow G723.1 in my sip.conf general config,
and from a bridge connection which gives silence,
I have progressed to the error message below,
and the call gets rejected.
help!!
Dave
ps. 217.168.168.49 : soft sipphone, i'm trying SJphone Pingel Instant
Expressa
[EMAIL
On Fri, 2003-06-06 at 15:32, Andy Powell wrote:
Tielman,
You can take a look at the quick and dirty guide I'm slowly putting together if you
like...
http://www.automated.it/guidetoasterisk.htm
I'd appreciate any feedback you have on it.. and if it helped
Andy
Excellent stuff
Newbie question alert!
I was just wondering how a receptionist phone would work with Asterisk.
(I've never had a real job, so I've never really looked at different
phone systems).
I have looked around on the internet and seen that you can purchase 24
line phones; how does that get connected?
On 06/06/2003 at 17:36 Patrick wrote:
Excellent stuff Andy. It was quite a disappointment that the document
stopped before explaining ..errr everything :) Look forward to learn how
to setup one-way conference and music on hold. Thanks for the guide so
far.
Regards,
Patrick
Glad it was of use
Just a quick query to find out if there are any other Asterisk users in
Colorado.
If you're out there, drop me a line off-list. I'd like to start a user
group if there is anyone else out there.
--Karl
--
Karl Putland [EMAIL PROTECTED]
___
What is the best
cost effective solution for a small office:
I need 3 FXS 2
FXO.
Can I hookup a
TDM400P and 2 X100P on the same computer?
Also, I saw some IP
phones for $25.99
http://www.wosmile.com/cgi-bin/view_store_item.cgi?pid=4424sid=5category=1
Can I use them with
asterisk? will
On Fri, 2003-06-06 at 16:09, Dante Alzamora wrote:
What is the best cost effective solution for a small office:
I need 3 FXS 2 FXO.
Can I hookup a TDM400P and 2 X100P on the same computer?
Also, I saw some IP phones for $25.99
Is there a website where you can order the Budgetone
100?
Thanks,
J.
__
Do you Yahoo!?
Yahoo! Calendar - Free online calendar with sync to Outlook(TM).
http://calendar.yahoo.com
___
Asterisk-Users mailing list
[EMAIL
53 matches
Mail list logo
