Thanks for all your comment.
I will release the source code but not for the moment. I really need to
clean the graphical part but right away I don't have enough time.
Regards,
Nicolas Bruxer
Lyle Giese wrote:
It does seem to work with ZAP channels and releasing the source would be a
great
Good day all
I want to tell asterisk that it should hangup a channel in a certain step
For example:
exten = s,5,Dial(SIP/302,25)
exten = s,6,Hangup
exten = s,7,Hangup(SIP/302)
What happens is that if someone calls into the pbx and hangs up before
it gets answered it still keeps on ringing on the
Peter,
Peter Landy wrote:
New to Asterisk so I am sure this has been answered before. I can
compile PWLIB and OpenH323 but when it comes to compiling asterisk-oh323
then I get all kinds of errors even though I have set the paths up in
the source files. I can attach the errors if it is useful. I
Hello,
On Sun, 21 Nov 2004, Tracy R Reed wrote:
On Sun, Nov 21, 2004 at 04:25:39PM -0800, Tony Vickers spake thusly:
What VOIP Phones is everyone using and why? Is the a common phone that
seems the work the best? Just wondering.
Snom - Good business class phones. Do they have any remote
Hi,
Didn't get any opinions on the log file I mailed onto the list over the
weekend so I am continuing to try and track the cause for the dropped
calls..
I have a feeling that its to do with IAX being way too sensitive when it
comes to packet loss.. Since it is going across the internet it
Tony Vickers wrote:
What VOIP Phones is everyone using and why? Is the a common phone that
seems the work the best? Just wondering.
I am using X-Lite and Siemens Optipoint 400s, simply because there is a
surplus of them at work from another installation. I've not had any
trouble with them so
I compiled the channel on usr/src/asterisk/channels/h323, which I believe is
the Nufone Channel.
Previously I did compile the PWLIB and OH323 packets.
Is that correct ?
Regards,
Jorge A.
-Mensaje original-
De: Paul Mahler [mailto:[EMAIL PROTECTED]
Enviado el: Sunday, November 21, 2004
Joseph wrote:
If I want to use IAX instead of SIP, do I need to get gateway that
support IAX.
Are there such gateways?
I plan to connect 3 to 4 standard phones via gateway with *
In addition I don't want to use SIP to setup VoIP. IAX is more suitable
for communication over firewall.
Joseph
Hi there,
Last week we discovered some problems using NAT/Routers and SIP Clients. We
had some work on our local network (exchanging routers) and so our clients
were temporarily offline without unregistering correctly. In our mysql
sipfriends table were several entries with the same IP (this is
Em Ter 09 Nov 2004 19:20, Richard Moore escreveu:
Hello list ,
I´m happy to announce our first stable production of a E100P
generic boards.
The prices are :
50 units = US$ 112,50/unit
50 units = US$ 98,10/unit
optional resources:
consulting :
32 US$ / server - You need provide
Peter,
If you have the lastest CVS version of asterisk(1.0.11) , and the latest
version of asterisk-oh323(0.7.0), it won't work.
What version of asterisk are you running? what version of oh323 are you
trying to compile?
K.
Peter,
Peter Landy wrote:
New to Asterisk so I am sure this has
Hi Jorge,
The oh323 channel and h323 channel by NuFone are different.
As far as your problem, this looks like a codec problem i had. Try to look
that way.
K.
- Original Message -
From: Jorge Alayon [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL
Yes I did. Does anyone have a working list of libraries and versions. I have
tried with different releases of H323 and they all give different errors.
Also is it necessary to compile the H323 under asterisk src/channels/H323
as this also bails on errors. The rest of my asterisk install works
Thank you,
I will need a SIP client with G723 and/or G.729 then. Do you know any sip
clients that do both ?
Regards,
Jorge A.
-Mensaje original-
De: kido noagbodji [mailto:[EMAIL PROTECTED]
Enviado el: Monday, November 22, 2004 8:42 AM
Para: Asterisk Users Mailing List -
Great program, thanks!
only one question: when i reboot my pc i can't see who is online, until the
sip user re-register their clients at the server. leon only seems to update
his online status when a sip client connects.
Jens
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Strange things are happening at my asterisk box :)
I've got asterisk setup to dail out with sip to my SIP provider.
I've got NO fromuser/fromdomain stuff setup in my sip.conf
When I place a call with my Siemens Optipoint 400 SIP phone everything is
allright, the From: header is stating the
I as a similar problem with this:
ignorepat = 9
exten = 9,1,Dial,Zap/g2
exten = 9,2,Congestion
What if I pressed 9, called a number, and hanged up before someone replies..
It happened with me more than once that the line is left open, waiting for
the other side to hangup (what if there is no
I guess it's a hoax. I did get a reply. They answered that they did not
have a website or any pictures of the boards they are producing.
The website was scheduled to be up last Wednesday but I don't get any
replies on e-mails either.
Remco
On Mon, 22 Nov 2004, Marcelo Pacheco wrote:
Em Ter 09
I am after something similar.
I want to be able to use 2 bonded ISDN BRI's and I am not sure what
hardware will run with asterisk?
Anyone got any ideas?
Cheers
David
Miroslav Nachev wrote:
Dear Bartosz,
Try this: http://www.junghanns.net/asterisk/page17.html
quadBRI PCI ISDN EUR 600,-
Hi,
this call is from? Zap channel, Capi channel or other channel? It is
possible that you don't detect well hangup from incoming channel.
Regards.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Altus Snyman
Enviado el: lunes, 22 de noviembre
Hmm, a bit closer, Asterisk seems to do the asterisk@ part only with non
numeric usernames, ie [EMAIL PROTECTED] stays [EMAIL PROTECTED] but
[EMAIL PROTECTED] turns into [EMAIL PROTECTED]
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens E. Versaevel
Yes I do have the latest CVS version and the 0.7.0 version of openH323. What
versions should I be using?
Regards
Peter Landy
-Original Message-
From: Michael Manousos [mailto:[EMAIL PROTECTED]
Sent: 22 November 2004 09:01
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
What VOIP Phones is everyone using and why? Is the a common phone that
seems the work the best? Just wondering.
The quality and features implemented in each of the voip phones varies
rather dramatically from one manufacturer to another. What works fine
in one account (with their
David Uzzell
I want to be able to use 2 bonded ISDN BRI's and I am not
sure what hardware will run with asterisk?
Anyone got any ideas?
I have a couple of customers with two HFC cards working on system access
(PTP) mode with no problems whatsoever. The cards have the major advantage
of
Good day all
I want to tell asterisk that it should hangup a channel in a certain step
For example: exten = s,5,Dial(SIP/302,25) exten = s,6,Hangup exten =
s,7,Hangup(SIP/302)
What happens is that if someone calls into the pbx and hangs up before
it gets answered it still keeps on
Having read through the Caller ID code, it appears that this is indeed what is
happening. The Caller ID code doesn't contain any logic to trigger a timeout
if no Caller ID data stream is found, or if a stream starts and does not
terminate.
The attached patch causes the Caller ID to timeout
Hi,
googd work,
could it be any chance to make it running on linux box ??
JFA
On Mon, 2004-11-22 at 09:16 +0100, Nicolas wrote:
Thanks for all your comment.
I will release the source code but not for the moment. I really need to
clean the graphical part but right away I don't have enough
On Thu, 18 Nov 2004 18:52:21 -0600, Jeb Campbell [EMAIL PROTECTED] wrote:
I'm replacing a Merlin for a client and they have a PagePal Intercom
that I would like to reuse.
Here is what I know about it:
It has a screw-down wires that goto rj-11 (This was told to me over the
phone) that went
Anybody used the above phone with asterisk
I have one working ok for calls, but having a problem with voice mail.
Using either the 'Voice mail function key' or dialing 88 (for my system)
just gets me to Call Terminated
Asterisk CLI shows the error message 'unable to get User name'
My
im getting:
Fatal error: Unknown function: mssql_get_last_message() in
/var/www/html/tikiwiki-1.8.2/lib/adodb/drivers/adodb-mssql.inc.php on
line 415
to the wiki..
Jason
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[EMAIL PROTECTED]
If your device supports Dial Off-Hook or PLAR Code, then you could send a
series of unique digits down the line to trigger the AGI script.
On Saturday 20 November 2004 12:52 pm, Michael Vogel wrote:
Michael Vogel schrieb:
Now I have got to find out how to make AGI play the dialtone
until a
Hi,
I don't kinow if I'm out of time but..
I in Swizerland (Geneva)
and use a E1 which is configured like this:
1) 120 ohmes on RJ45 (which is a standard)
2) digium cards support both but I have CRC4 on
3) sure it's handled two way full
IF I can help you in anyway don't hesitate to contact me
Hello,
An hoax is a code that pretend to be malicius but in fact isn't (usualy
a virus that do nothiing bad just saying that is there)
ragards
JFA
On Mon, 2004-11-22 at 09:33 -0200, Marcelo Pacheco wrote:
Em Ter 09 Nov 2004 19:20, Richard Moore escreveu:
Hello list ,
Im happy to
Brian Wilkins schrieb:
If your device supports Dial Off-Hook or PLAR Code, then you could send a
series of unique digits down the line to trigger the AGI script.
Now everything works. So I don't need any tricks. Yesterday I wrote a
simple dialplan that suits my needs.
Yesterday I also wrote a
Hi,
Il giorno lun, 22-11-2004 alle 08:49 -0500, Jason p ha scritto:
Fatal error: Unknown function: mssql_get_last_message() in
/var/www/html/tikiwiki-1.8.2/lib/adodb/drivers/adodb-mssql.inc.php on
also here...perhaps they're switching away from mssql ? :)
Matteo
--
Matteo Brancaleoni
System
What you describe is a computer hoax.
A look in a dictionary will give you this:
Main Entry: hoax
Pronunciation: 'hOks
Function: transitive verb
Etymology: probably contraction of hocus
Date: circa 1796
: to trick into believing or accepting as genuine something false and often
preposterous
Thanks for the link
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*bump*
(B
(Bwoops..this isn't a forum...
(B
(Bbut anyway, this is a good question. We will soon have somewhere in the 500
(BSIP users range and if I can have 1 machine (SER) handle all the
(Bregistration (hopefully out of a database) that will defiantly reduce the
(Bload on my asterisk
In recent times I have seen a few posts which
describe how to use ztmonitor to set the rxgain and txgain parameters on an FXO
channel.
The starting point is to call a '102 milliwatt test
number'
Does anybody know of such a facility in the
UK?
Ian
On Sun, 21 Nov 2004 18:54:25 +0500, khurram bhatti
[EMAIL PROTECTED] wrote:
I'm trying to connect * server from diax 0.9.8c client and * outputs this
errors on CLI
Nov 21 18:59:59 NOTICE[7316]: chan_iax2.c:5742 socket_read: Rejected connect
attempt from 192.168.0.4, requested/capability
From what I've been reading about the callprogress option, it seems like it
will work properly only with a T1 or PRI in the US. Is that correct or are
there still issues with call progress detection even if those qualifications
are met?
Thanks,
Shaun Tierney
I am quite interested in this as well. I didn't realize registrations are
the #1 cause of load on an asterisk server, we haven't gotten to that kind
of usage just yet.
People were having problems with compiling Asterisk on a hacked Linksys
WRT54G, issues with compiling against uClibc and some
On Sun, 21 Nov 2004 19:50:36 -, Kevin Brennan
[EMAIL PROTECTED] wrote:
I am planning to configure * box A with PSTN interface to route faxes to *
box B (running spandsp) over TDMoE. I am using 2xGb bonded NIC's for
connection between servers.
Was wondering
- does anybody have experience
Hi to everybody,
is it possible to use ISDN Call Deflection with a ZapHFC card?
Regards
Bastian
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Hello, I am a newbie with asterisk; I´ve searching the mailinglist,
www.voip-info.org, isdn4linux web... But I don´t know which isdn card to
buy.
I want the * box to be able to send faxes, and obviusly to send and receive
calls.
1) What do you recomend me?
2) Would AVM ISDN Fritz Card PCI V2.0
Hi,
You are true for the moment, I only listen to messages and don't
request anything to asterisk. So I only manage registration when a sip
phone register or unregister while the leon is already launched.
Two possible solutions:
-Havin a server that remember states which I want to avoid
Hi,
Sorry it is not planned.
Regards,
Nicolas
genuix wrote:
Hi,
googd work,
could it be any chance to make it running on linux box ??
JFA
On Mon, 2004-11-22 at 09:16 +0100, Nicolas wrote:
Thanks for all your comment.
I will release the source code but not for the moment. I really need to
Message: 4
Date: Sun, 21 Nov 2004 17:56:10 -0800
From: Paul Mahler [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;
On Mon, 22 Nov 2004, Jason Williams wrote:
I recommend you use Iax trunking rather than TDMoE this would scale better.
Using iax trunking will also loose the advantage of being tdm all the way,
i.e. low latancies. If the rest of the setup is tdm there is a lot of
value in not going to voip
On Sun, Nov 21, 2004 at 05:30:13PM +0200, Pertti Pikkarainen wrote:
However I would use a more specific path for a web-server ;-) Something
like:
option tftp-server-name http://192.168.0.9/snom/snom200.htm
But for the snom 190 tftp-server-name in dhcp config will set
update_server. The
I have no experience with faxes * at all but the AVM ISDN Fritz Card
PCI V2.0 works very well for me (with SuSE 9.1) and I found them easy to
get on ebay.de
Derek
Rubens Sanchez wrote:
Hello, I am a newbie with asterisk; I´ve searching the mailinglist,
www.voip-info.org, isdn4linux web... But
Hello,
I'm a new Asterisk user and I hope I haven't missed something, but I
can't seem to find an answer to this issue. I have a Cisco SIP
gateway terminating calls into a 7960 phone. The issue I would like to
fix is if I have an incoming call without an ANI, such as directly from
my TDM phone
Has anyone tried out the Linksys RT31P2
with Asterisk? Seems like a really great solution for remote users... even
supports QoS. Too bad it doesn't also have VPN functionality built
in.
Here's a link to the product:
http://www.linksys.com/products/product.asp?prid=652scid=29
Hi there,
How do i setup asterisk, so that in the CDR's is only the time, which the
line actually was connected? Not the time, the line was up, but the time the
user was able to talk to another user.
Thanks in advance,
Carsten
___
Asterisk-Users
Has anyone been successful interfacing call manager and Asterisk?
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Altus Snyman wrote:
Good day all
I want to tell asterisk that it should hangup a channel in a certain step
For example:
exten = s,5,Dial(SIP/302,25)
exten = s,6,Hangup
exten = s,7,Hangup(SIP/302)
What happens is that if someone calls into the pbx and hangs up before
it gets answered it still
If anyone finds the generic version of this available (i.e., not locked to
Vonage), please advise the list of where.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 22 Nov 2004 [EMAIL PROTECTED] wrote:
Has anyone tried out the Linksys RT31P2 with Asterisk?
Try George at www.netvoice.ca
- Original Message -
From: Joseph [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, November 21, 2004 9:26 PM
Subject: [Asterisk-Users] SPA-841 / SPA-2100 Canadian Distributor
Does anybody know
WipeOut wrote:
Hi,
Didn't get any opinions on the log file I mailed onto the list over
the weekend so I am continuing to try and track the cause for the
dropped calls..
I have a feeling that its to do with IAX being way too sensitive when
it comes to packet loss.. Since it is going across the
Peter Landy wrote:
Yes I did. Does anyone have a working list of libraries and versions. I have
tried with different releases of H323 and they all give different errors.
Also is it necessary to compile the H323 under asterisk src/channels/H323
as this also bails on errors. The rest of my asterisk
[EMAIL PROTECTED] is believed to have said:
Hello, I am a newbie with asterisk; I¥ve searching the mailinglist,
www.voip-info.org, isdn4linux web... But I don¥t know which isdn card to
buy.
I want the * box to be able to send faxes, and obviusly to send and receive
calls.
1) What do you recomend
On November 22, 2004 10:47 am, [EMAIL PROTECTED] wrote:
Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really
great solution for remote users... even supports QoS. Too bad it doesn't
also have VPN functionality built in.
How well do these Geode and ARM-based systems
On November 19, 2004 05:17 pm, FuturaHost.Com Lists wrote:
Yes and no would suffice, so we can close this without a talk long a
year, and without someones forcing their point of view to others.
Sorry but yes or no does not suffice because the very next post will be
Why?
-A.
Hi Will,
snip
Having read through the Caller ID code, it appears that this is indeed
what is happening. The Caller ID code doesn't contain any logic to
trigger a timeout if no Caller ID data stream is found, or if a stream
starts and does not terminate.
/snip
I submitted a patch for this
Shaun Tierney wrote:
From what I've been reading about the callprogress option, it seems like it
will work properly only with a T1 or PRI in the US. Is that correct or are
there still issues with call progress detection even if those qualifications
are met?
If you ask me it doesnt' work well
Thank you, I will see into it.
Regards,
Jorge A.
-Mensaje original-
De: Paul Davidson [mailto:[EMAIL PROTECTED]
Enviado el: Monday, November 22, 2004 12:12 PM
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper
Message: 4
Date: Sun, 21 Nov 2004
-Original Message-
From: Stefano Finetti [mailto:[EMAIL PROTECTED]
Sent: November 19, 2004 6:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unpredictables Hangups
Michael,
I just check'd my kernel configuration...
I have APIC
Ok, so if I turn callprogress off, and try to connect a call which is
bridged between an incoming line and an outgoing line, will it treat the
call as being answered once it is bridged or once it is actually answered on
the outgoing T1 trunk?
Thanks,
Shaun
-Original Message-
From:
snip
How well do these Geode and ARM-based systems
handle VPN anyway? I would have
figured you would want a decent processor to handle more than maybe one
or
two clients.
/snip
These are really designed for home use. I
use the BEHVP41 to keep ~ 10 VPN tunnels open. Works great. Not sure at
Yes, I have both Call Manager and Call Manager Express integrated with *.
Prior to Call Manager 4.0 you would need to perform an H.323 integration
with *. As of CM 4.0 Cisco supports SIP trunking so this would be the
preferred method of integration. This config is on http://www.voip-info.org
Andrew Kohlsmith wrote:
How well do these Geode and ARM-based systems handle VPN anyway? I would have
figured you would want a decent processor to handle more than maybe one or
two clients.
The CPU is only a limitation for a VPN if the pipe the VPN is running
over is large/wide. These devices
I think the USB IP Phone adaptor is a S100U - I found the TigerJet
website/products by reading the chip inside a S100U that I purchased at
digium and they look identical - but dont trust me on this - I didn't
buy one from TigerJet direct.
Derek
Michael Vogel wrote:
Derek Conniffe schrieb:
Re:
Chris Olson wrote:
Hello,
I have firefly installed and it is somewhat working. It is registering
with my Asterisk server and I can call out, but I receive no audio
coming into Firefly. From the Asterisk end, everything looks OK with
the call, just no audio is being received on the Firefly end.
On Fri, 2004-11-19 at 18:01, Kevin wrote:
I have applied the revised patch. After working through the steps to
follow (I think there may be another mistake in the steps) I get a busy
when calling out:
-- Got SIP response 404 Not Found back from 147.135.0.128
--
Has this worked finally? Can you send me the configs if they indeed
have been working.
Seshu kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steffen
Koepf
Sent: Friday, November 19, 2004 8:40 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
I'm losing call files in /var/spool/asterisk/outgoing because * isn't
able to detect the busy signal. The call file looks like this:
Channel: Zap/g2/3036701917
MaxRetries: 1000
RetryTime: 60
WaitTime: 45
Application: TxFAX
Data: filename.tiff|caller
Using the |caller parameter, TxFax injects
Anybody using Siemens optiPoint 300 H.323 phones? I saw a few references to
them in the archives of this list, and the Wiki seems to be down.
I have a chance to pick up a bunch of these, cheap.
Questions:
* Asterisk support?
* What sort of power supplies will they need? The bunch I am looking at
On Monday 22 November 2004 16:01, Stefan Tichy wrote:
On Sun, Nov 21, 2004 at 05:30:13PM +0200, Pertti Pikkarainen wrote:
However I would use a more specific path for a web-server ;-) Something
like:
option tftp-server-name http://192.168.0.9/snom/snom200.htm
But for the snom 190
Any ideas on this warning? If I call this number, sometimes I get this error
and sometimes the call goes thru fine. Why would it work sometimes?
-- Executing Goto(SIP/3044-8d49, cytel-outgoing|915124512424|1) in
new stack
-- Goto (cytel-outgoing,915124512424,1)
-- Executing
I'm losing call files in /var/spool/asterisk/outgoing because * isn't
able to detect the busy signal. The call file looks like this:
Channel: Zap/g2/3036701917
MaxRetries: 1000
RetryTime: 60
WaitTime: 45
Application: TxFAX
Data: filename.tiff|caller
Using the |caller parameter, TxFax injects
Michael Welter wrote:
Using the |caller parameter, TxFax injects the fax tone (CNG) onto the
line. With the CNG tone, asterisk is unable to detect the busy tones.
If I were to remove |caller then the receiving station wouldn't
receive the CNG tone and possibly not direct the call to the fax
Matthew Boehm wrote:
-- Executing Dial(SIP/3044-8d49, ZAP/g1/15124512424|60|t) in new
stack
Nov 22 10:59:32 WARNING[1126867776]: chan_zap.c:4159 zt_write: Cannot handle
frames in 256 format
show codecs in the Asterisk CLI will tell you the number of each codec.
If you want to use G729 and the
Hi,
I've been experimenting with an IPv4 and IPv6 VoIP setup using SER.
I'm using Asterisk for voicemail, etc. but as this only works for
IPv4, I had to do a number of tricks to get it going for IPv6 phones.
I was wondering whether there is any interest or plans in the pipeline
to have Asterisk
I guess I should have mentioned that I have 10 codecs:
0/0 encoders/decoders of 10 licensed channels are currently in use
Any other ideas?
-Matthew
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Brian McCrary wrote:
Hello,
I'm a new Asterisk user and I hope I haven't missed something, but I
can't seem to find an answer to this issue. I have a Cisco SIP
gateway terminating calls into a 7960 phone. The issue I would like to
fix is if I have an incoming call without an ANI, such as
All,
I was wondering if it is possible to use the manager api to
stop a "stream file" agi command for a channel.
Either through posting a DTMF digit to the channel or
something like that - or a cleaner way also.
My AGI cannot cancel the playing of a "Stream file" command
unless the user
On Mon, Nov 22, 2004 at 04:09:21PM -, Ian D. Wlloughby wrote:
I submitted a patch for this which was included in the the CVS build of
the 19th of November.
See bug http://bugs.digium.com/bug_view_page.php?bug_id=0002909
Ian,
The code I am running includes this patch already (I checked it
Hello,
I am trying a couple of days before to set up asterisk to redirects an
incoming call if the extension dialed is busy without success.
I just try to use 'Gotoif' command, with bad luck, it can't do what i want.
Anybody could helpme?
ani clue will be appreciated.
Regards.
Ismael.
Ian,
The code I am running includes this patch already (I checked it out
from CVS on the 21st), and it does not seem to resolve the problem on my
line.
The patch I have posted earlier today, which causes the Caller-ID code
to abort after 15 seconds, does resolve the problem.
Hmmm, it should
Read up on SetGroup and CheckGroup.
On Nov 22, 2004, at 9:57 AM, ismaelg wrote:
Hello,
I am trying a couple of days before to set up asterisk to redirects an
incoming call if the extension dialed is busy without success.
I just try to use 'Gotoif' command, with bad luck, it can't do what i
On Mon, 2004-11-22 at 18:57 +0100, ismaelg wrote:
Hello,
I am trying a couple of days before to set up asterisk to redirects an
incoming call if the extension dialed is busy without success.
I just try to use 'Gotoif' command, with bad luck, it can't do what i want.
Anybody could
Hi all,
I've got one of those cool free incoming IPKall phone numbers from
www.ipkall.com. These numbers just connect to the SIP proxy of your
choice, they default to Frreworld Dialup. You can use them with your own
sip proxy on asterisk. My config for this is below.
The trouble I'm having
I have the Cisco 7960 SIP version 7.3 phone.
When someone calls in I cannot always hear that person.
They can hear me though. (The ear piece is DEAD quite like it
is muted or something - no noise at all).
This never happens with the other 4 grandstream SIP phones I have.
Is there a problem
Using 7.3 here on a 7960 and no problems.
Matthew
- Original Message -
From: Jerry Geis [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 22, 2004 12:51 PM
Subject: [Asterisk-Users] Cisco 7960 version 7.3 SIP not always able to
hearcalling person
I have the Cisco 7960 SIP
Rob Emanuele [EMAIL PROTECTED] wrote:
I've got one of those cool free incoming IPKall phone numbers from
www.ipkall.com. These numbers just connect to the SIP proxy of your
choice, they default to Frreworld Dialup. You can use them with your own
sip proxy on asterisk. My config for this is
On Fri, 2004-11-19 at 15:31 -0800, Erik Espinoza wrote:
I can't seem to get this device to grab an ip from dhcp. We have a
working dhcp server (unfortunately it is on Windows), but I don't show
any leases requested by the iaxy.
Anyone have any ideas?
The ethernet and phone lines are
Rob Emanuele [EMAIL PROTECTED] wrote:
I've got one of those cool free incoming IPKall phone numbers from
www.ipkall.com. These numbers just connect to the SIP proxy of your
choice, they default to Frreworld Dialup. You can use them with your
own
sip proxy on asterisk. My config for this
Well, it seems that Zap cannot do 729 at all:
channels/chan_zap.c (line 4156):
if ((frame-subclass != AST_FORMAT_SLINEAR)
(frame-subclass != AST_FORMAT_ULAW)
(frame-subclass != AST_FORMAT_ALAW)) {
ast_log(LOG_WARNING, Cannot handle frames in %d format\n,
Hello,
I have a similar issue with the PingTel xpressa: audio is not sent from the
phone to *. Has anyone else experienced it?
Best,
Alessandro
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Monday, November 22, 2004 10:59 AM
Matthew Boehm wrote:
Well, it seems that Zap cannot do 729 at all:
channels/chan_zap.c (line 4156):
if ((frame-subclass != AST_FORMAT_SLINEAR)
(frame-subclass != AST_FORMAT_ULAW)
(frame-subclass != AST_FORMAT_ALAW)) {
ast_log(LOG_WARNING, Cannot handle frames in %d
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