Title: IAX2 Problem and Question
Dear Asterisk Users.
I have been setting up IAX between two servers, one in the USA and the other in UK so that I can pass help desk and general calls from one call center to another.
I seem to be having an issue. When I set up IAX between my two servers I
Good day all
So I've installed asterisk with my openline4 card and I've setup sip and
I can do sip on the local network,we are using soft clients,x-lite.
But...
When a call comes in from the outside(PSTN) and the you dial the
extension it forwards the call the the client and you see incoming call
Welcome to the Asterisk users community!
It's great to have you participating in this Open Source project
- building an Open Source PBX. Here are a few things to know and
remember while working with the project.
** The mailing list is growing
The lead
Title: Message
here
you go :)
http://bugs.digium.com/bug_view_page.php?bug_id=214
Ta
SJ
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
KirklandSent: 05 April 2004 04:25To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] New Call
Use the shell command ! to exit to shell.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Parlee
Sent: Sunday, April 04, 2004 3:56 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unabled to exit console
No matter what I try, Asterisk won't let
On Wed, 31 Mar 2004, Senad Jordanovic wrote:
Angus Berry wrote:
A quick search on eBay turned up this 4 port FXO external box for
US$299:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3087347715category=51279
...anyone know if it's compatible with Asterisk?
Yes.. I can confirm I had
Fran Boon wrote:
Gavin Hamill wrote:
I'm using Mozilla 1.7a installed from a tarball. The needed libraries
are just there:
You've answered your own question. You installed Mozilla from a
tarball. RPM therefore doesn't know about it. You need to install a
recent Mozilla RPM :)
Why do I need to
Brian Cuthie wrote:
Let's say that I have a call coming in to Asterisk through a TDM400P and
going out through SIP to someone on the Internet. Is there any
configuration option that would allow me to do silence suppression on
the RTP stream generated by Asterisk on behalf of the TDM400P
jc wrote:
Use the shell command ! to exit to shell.
Use screen...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your
Larry Keyes wrote:
Hi...I've got two Grandstream phones attached to my Asterisk on the same
subnet. The phones have fixed IP addresses. Asterisk is generated an error
for one of them only, even though both appear to be registered correctly.
The current state of the sip.conf is included below.
Hello,
Asterisk in FreeBSD ports is currently FORBIDDEN due to security issues
raised in pwlib (H323). As I just want to test Asterisk internally at
this point I commented out the FORBIDDENs and compiled it with no problems.
Unfortunately though, I can't seem to get any SIP softphones to register
Richard Airlie wrote:
At this stage, and someone please confirm that Asterisk is
really working on FreeBSD ? :)
Yes, it's working with some limitations.
See http://www.voip-info.org/tiki-index.php?page=Asterisk%20freebsd
/O
___
Asterisk-Users mailing
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Olle E. Johansson
Brian Cuthie wrote:
Let's say that I have a call coming in to Asterisk through
a TDM400P
and going out through SIP to someone on the Internet. Is there any
On Mon, 5 Apr 2004, Martin Mielke wrote:
Why do I need to install from RPM when I already included the Mozilla
lib directories in /etc/ld.so.conf and issued a 'ldconfig' command? The
system should know where to look for the needed libraries already...
The system might (depending on how you
On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote:
Richard Airlie wrote:
At this stage, and someone please confirm that Asterisk is
really working on FreeBSD ? :)
Yes, it's working with some limitations.
See http://www.voip-info.org/tiki-index.php?page=Asterisk%20freebsd
Hi !!
I know that a conference room can be made infinitely.
but, I think that there is actually a limit.
For example, how many conference rooms can be made from CPU 866 [MHz] and
RAM 256 [MB]?
Is there any person who tried someone?
I am studying MeetMe now.
Hi *ers,
I recently got an Email from Redhat about the dropping of support for Redhat
9 on the 30 of April and that Fedora Project is the recommended future,
otherwise, RedHat enterprise ($$$).
Considering this, I would like some feed back on the Fedora Project from
users who may be using it,
Richard Airlie wrote:
On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote:
Richard Airlie wrote:
At this stage, and someone please confirm that Asterisk is
really working on FreeBSD ? :)
Yes, it's working with some limitations.
See
At 8:34 AM -0400 on 4/5/04, Brian Cuthie wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Olle E. Johansson
Brian Cuthie wrote:
Let's say that I have a call coming in to Asterisk through
a TDM400P
and going out through SIP to someone on
|From: James Gardiner
| Hi *ers,
| I recently got an Email from Redhat about the dropping of support for
Redhat
| 9 on the 30 of April and that Fedora Project is the recommended future,
| otherwise, RedHat enterprise ($$$).
| Considering this, I would like some feed back on the Fedora Project from
I am only just starting out with * myself, but believe it or not I had
the same problems not more than a couple of days ago.
1) With the X-Lite clients I was able to connect a call amongst them,
but unable to hear a thing. (Same problem I suspect). The problem
ended up being that the * server
There was a question about this earlier. I had a similar problem and
fixed it by specifying the audio protocol to be used in the general
section of the sip.conf.
-Original Message-
From: Altus Snyman [mailto:[EMAIL PROTECTED]
Sent: Monday, April 05, 2004 3:52 AM
To: asterisk
Subject:
James Gardiner wrote:
Hi *ers,
I recently got an Email from Redhat about the dropping of support for Redhat
9 on the 30 of April and that Fedora Project is the recommended future,
otherwise, RedHat enterprise ($$$).
Yup, this has been coming up for a while now..
Considering this, I would like
Hi *ers,
I recently got an Email from Redhat about the dropping of support for
Redhat
9 on the 30 of April and that Fedora Project is the recommended future,
otherwise, RedHat enterprise ($$$).
Considering this, I would like some feed back on the Fedora Project from
users who may be using
two wrote:
Hi !!
I know that a conference room can be made infinitely.
but, I think that there is actually a limit.
For example, how many conference rooms can be made from CPU 866
[MHz] and RAM 256 [MB]? Is there any person who tried someone? I am
studying MeetMe now. Please tell me a
We had an issues with an Intel Zero Channel hardware RAID controller
that wouldn't allow us to install either Fedora Core 1 or 2, so we
couldn't test with *. Given that we didn't try to convert our 9 to
Fedora, either. We got it running great under RH 9.
HTH,
Ryan Thrash
On Apr 5, 2004, at
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Glen Ford
Sent: Friday, April 02, 2004 2:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] avaya and linux
Does anyone know if avaya voip product is running linux under
the hood?
Yes. The
Hi,
My asterisk fails and stops after running the reload command ~20 times (I'm
testing) - is this a kown problem ?
Therefor I wil reload only sip, extensions and iax, it works with sip and
extensions, but it seem that there are no reload for iax - or what ?
--
mvh. Hans-Henrik Andresen
Hi !!
Thank you for teaching!!
A question is changed for a while.
please tell me the information that the conference room was
able to be made how many, by which spec.
English cannot be used well and it is pardon!!
English is also under study.
On Mon, Apr 05, 2004 at 12:49:24AM -0400, Shad Mortazavi wrote:
Dear Asterisk Users.
I have been setting up IAX between two servers, one in the USA and the other
in UK so that I can pass help desk and general calls from one call center to
another.
I seem to be having an issue. When I set
Can anyone point me to some sample Cisco QoS configurations suitable for
IAX2? I've looked through Cisco's site, and get overwhelmed with the level
of documentation (too much of a good thing).
My PSTN gateway and PBX (both *) are connected via 2xT1 (per-packet load
balancing) between a Cisco
Personally, I think VAD is a great service, as well as comfort noise
generation to disguise when VAD is working. I'll always encourage
methods that reduce bandwidth. Most major developers on Asterisk
consider these technologies of low concern since their bandwidth is
unlimited, as they
Are you in control of both sides? What routing protocols are you using?
Simply using Cisco CAR can help, but not a total solution. Are the 2 T1's
carried by an ISP? Or are these private T's?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Troy Settle
Olle E. Johansson wrote:
Richard Airlie wrote:
On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote:
Richard Airlie wrote:
At this stage, and someone please confirm that Asterisk is
really working on FreeBSD ? :)
Yes, it's working with some limitations.
See
Hi!
I am running an * 0.7.2 on an X86 debian stable 2.4.25 (with
backports.org). The HW I am using is Digium's E100P on an HP DL 380.
Quite often it crashes, e.g. after a call has finished. Below some logs
form the * Console as well as from the /var/log/asterisk/messages
(Replaced some stuff
Hi Troy,
Troy Settle wrote:
Can anyone point me to some sample Cisco QoS configurations suitable for
IAX2? I've looked through Cisco's site, and get overwhelmed with the level
of documentation (too much of a good thing).
Take a look at this and see if you can use it for IAX2 as well:
Andrey McRory built a RPM dist for * but I can't seem to find it anywhere..
Any hints where I might be able to find this package that has matching
kernel?
Thanks,
Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I regret that I've only used MeetMe a few times, and only up to two users.
Perhaps others that are using MeetMe could comment on the number of
concurrent conferences and total users they have asterisk running with. The
specs of the systems involved would be most helpful.
If this is on the wiki,
Christopher C. Howard wrote:
Andrey McRory built a RPM dist for * but I can't seem to find it
anywhere.. Any hints where I might be able to find this package that
has matching kernel?
This is what I found for rpm. http://www.voip-info.org/wiki-Asterisk+RPM
Hope this helps.
Thanks,
Chris
I regret that I've only used MeetMe a few times, and only up to two users.
Perhaps others that are using MeetMe could comment on the number of
concurrent conferences and total users they have asterisk running with.
The
specs of the systems involved would be most helpful.
I have set up a
quote who=Andrew Thompson
I regret that I've only used MeetMe a few times, and only up to two users.
Well, the problem with giving general stats, is that it REALLY depends on the
exact environment.
Key points: (on a server dedicated for conferences only)
o number of channels
o types of
Thank you James for reply.
Conole does not print any messages.
When I trace SIP messages I can see that invitation is sent, and then it
call is explicitly hung up. The phone starts to ring for a second and then
goes quiet. The same thing happens if I originate on a Zap channel.
On Zap channel
I am having an issue with Callerid (INBOUND). I have a system set up
with 4 companies sitting behind the system. On all of the companies
except of one of them, it displays callerid withh 'asterisk'. The other
company displays the callerid of the person calling.
Zapata.conf
[channels]
I have made bri-stuff.0.0.2rc19 to work (I think) but I can not achieve
any in-dialing nor I can dial out;
this is what I have from pri intense debug span 1 command
--
*CLI pri intense debug span 1
Enabled EXTENSIVE debugging on span 1
-- Executing Playback(SIP/201-a862,
I have an intermittent problem with the one FXS line that I have. On
most calls, the first ~5 seconds of the call has a loud buzzing noise
on the line. After 5 seconds or so, it fades off to nothing, and the
sound quality is great. Searching for buzzing on the list doesn't
give a whole lot
Having just returned from four days at the VON show in Santa Clara, I
thought I would submit a highlights message. I hope others who attended
the show will take the opportunity to add, as there was far more to see than
I can cover on my own.
[VoIP IS BIG]
First, I have to say that VoIP is BIG.
Haven't seen this, but I do hear a loud click about 5 seconds into any call
involving a TDM400P port. Seems like something might not be quite right with
the Zap driver.
-brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Scott Laird
Sent:
Steven Sokol wrote:
Having just returned from four days at the VON show in Santa Clara, I
thought I would submit a highlights message. I hope others who attended
the show will take the opportunity to add, as there was far more to see than
I can cover on my own.
Thank you for a good report!
Packetwest Communcations provides local IAX termination service in Seattle.
I use it locally for a small Asterisk setup and they provide me with DID's
in the 206 NPA. They also provide outbound long-distance at rates similar
to NuFone. I've had a really good experience with service quality and
X-Analitica - MD-MailScanner-OpenProtect-Information: Please contact the ISP for more
information
X-Analitica - MD-MailScanner-OpenProtect: Found to be clean
X-MailScanner-MCPCheck:
is it already inside * 0.7.2?
El lun, 05 de 04 de 2004 a las 03:21, Senad Jordanovic escribi:
here you go :)
On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote:
Members of the IETF added information on the to-be-standardized
standard,
meaning that SIP with TLS over TCP will be mandatory. We need to start
working
on TCP and TLS support.
Could someone explain to me why anyone in their right mind would
I found some posts regarding this issue dating of September 2003, but no
real answer.
The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I
need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help
migration.
Is there an existing format/codec for this? If not,
On Apr 5, 2004, at 12:18 PM, Mark Hagler wrote:
Packetwest Communcations provides local IAX termination service in
Seattle.
I use it locally for a small Asterisk setup and they provide me with
DID's
in the 206 NPA. They also provide outbound long-distance at rates
similar
to NuFone. I've
On Mon, 5 Apr 2004, Scott Laird wrote:
Could someone explain to me why anyone in their right mind would ever
want to run VoIP (or any lossy real-time data) over TCP? Unless I'm
missing something, the effects of packet loss would be almost perfectly
pessimal. Every time you lose a
On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote:
Members of the IETF added information on the to-be-standardized
standard,
meaning that SIP with TLS over TCP will be mandatory. We need to start
working
on TCP and TLS support.
Could someone explain to me why anyone in their right
On Apr 5, 2004, at 12:34 PM, James Golovich wrote:
On Mon, 5 Apr 2004, Scott Laird wrote:
Could someone explain to me why anyone in their right mind would ever
want to run VoIP (or any lossy real-time data) over TCP? Unless I'm
missing something, the effects of packet loss would be almost
Asterisk - MD wrote:
X-Analitica - MD-MailScanner-OpenProtect-Information: Please contact
the ISP for more information X-Analitica -
MD-MailScanner-OpenProtect: Found to be clean X-MailScanner-MCPCheck:
is it already inside * 0.7.2?
Yap...
___
Scott Laird [EMAIL PROTECTED] wrote:
On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote:
Members of the IETF added information on the to-be-standardized
standard,
meaning that SIP with TLS over TCP will be mandatory. We need to start
working
on TCP and TLS support.
Could someone
Scott Laird wrote:
On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote:
Members of the IETF added information on the to-be-standardized standard,
meaning that SIP with TLS over TCP will be mandatory. We need to start
working
on TCP and TLS support.
Could someone explain to me why anyone in
James Golovich wrote:
On Mon, 5 Apr 2004, Scott Laird wrote:
Could someone explain to me why anyone in their right mind would ever
want to run VoIP (or any lossy real-time data) over TCP? Unless I'm
missing something, the effects of packet loss would be almost perfectly
pessimal. Every time
Steven Sokol wrote:
I think Olle Johansson took pictures of the event.
They may already be on the WiKi in fact.
I've uploaded the pictures without editing at
http://www.voip-forum.com/asterisk/von2004/index.htm
Enjoy!
/O
___
Asterisk-Users mailing
You can also take a look at the following URL:
http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_command_ref
erence_chapter09186a0080087f26.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Troy Settle
Sent: Monday, April 05, 2004
Hi,
We are using Nufone as our voip provider and it is working fine except for the problems i mentioned in my email.
Thanks
Owais Bin Zuber"James H. Thompson" [EMAIL PROTECTED] wrote:
Just curious - was wondering who you are using as your VOIP provider and how its working out?
Thanks
Jim
On Apr 5, 2004, at 12:49 PM, Olle E. Johansson wrote:
SRTP protects RTP/UDP media with encryption.
There are concerns that sending positioning within SIP/UDP will reveal
private detailes, like position. Hence the encryption requirement.
The position data needs to be given by the ISP in DHCP
K...maybe this was stated earlier in the conversation...but what's the
deal with the phone? Or was this phone just being carried around by
everyone and ripped apart?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Monday, April
Hi there,
Is there anyway to make the RTP data flow directly a SIP phone
and a H323 phone through the oh323 or chan_h323 modules? Something like waht
the canreinvite = yes option inside the sip.conf does for SIP to SIP calls.
thanks,
Pablo Salinas
At 01:44 PM 4/5/2004, you wrote:
Having just returned from four days at the VON show in Santa Clara, I
thought I would submit a highlights message. I hope others who attended
the show will take the opportunity to add, as there was far more to see than
I can cover on my own.
Was there any
Scott Laird wrote:
On Apr 5, 2004, at 12:49 PM, Olle E. Johansson wrote:
SRTP protects RTP/UDP media with encryption.
There are concerns that sending positioning within SIP/UDP will reveal
private detailes, like position. Hence the encryption requirement.
The position data needs to be given by
Mark Messmore, Technical Support, University Telcom Inc. wrote:
K...maybe this was stated earlier in the conversation...but what's the
deal with the phone? Or was this phone just being carried around by
everyone and ripped apart?
Old Bell Phone + IAXy + 802.11b card + Batteries = Homemade
Scott Laird wrote:
2. Is anyone working on SRTP for Asterisk? Are there any SRTP clients
out there?
Checked again, the vovida.org and the sourceforge one are the same.
And here's the good news: THey're using a BSD license.
That means we can incorporate this library into Asterisk without a
Hello i was wondering how i can change the IP
address information for my Asterisk box, IP addy, Gateway, DNS.
I have a smoothwall router that i am using and i am
tring to put the Asterisk box on the orange interface so if anyone can help me
please i can use it.
Thanks alot
William Ray
Was there any aggressive pricing given for nationwide voip LD?
Level3 had several products, one they called Enhanced which was supposed
to also include E911 service. They quoted me about $.01 per minute
inbound or outbound nation wide. They said they support the top 300
cities in the US and, of
I am trying to get dtmf digits to pass from a SNOM 200 through * to a Cisco
AS5300. I have setup the cisco gateway and the sip.conf file to use rfc2833
and I have disabled inband dtmf on the snom 200. Unfortunately, the digits
are still not being passed. Something tells me that I am missing
Title: Disambiguating incoming IAXTel calls
I have two 1-700 numbers from IAXTel. Both get registered from the same Asterisk server. I can make and receive calls on each without any difficulty. What I can't figure out how to do is route the incoming calls differently based on which 1-700
I have the voicemail setup working in that I get the MWI and it emails the
message correctly. When I pressed the MWI button on my SNOM 200, it dials
into the voicemail system and prompts me for a mailbox and password. I know
there is a way to automatically connect directly into the mailbox via the
Their base rate is $35/mo per peer (single call transit at any given moment)
and this provides unlimited local and inbound calling. If you are
connecting a PBX and need 1 voice path at any given moment you can discuss
different pricing arrangements for your needs.
DID numbers are 15
I have my server configured to send to send all PSTN traffic to my Cisco
AS5300 gateway via SIP. I use the following line in the extensions.conf file
to accomplish this:
exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],240,T)
Unfortunately, when I removed the T from the end of the statement, the
On Apr 5, 2004, at 1:57 PM, Brian Rathman wrote:
I have the voicemail setup working in that I get the MWI and it emails
the
message correctly. When I pressed the MWI button on my SNOM 200, it
dials
into the voicemail system and prompts me for a mailbox and password. I
know
there is a way to
I use something like this:
exten = 8500,1,Ringing
exten = 8500,2,Wait,1
exten = 8500,3,VoicemailMain(s${CALLERIDNUM})
Basically, this rings the phone for once second (thus setting up the audio
path), then goes to voicemail without requiring the password. Leave out the
's' to have VM prompt for
Bob Klepfer wrote:
Mark Messmore, Technical Support, University Telcom Inc. wrote:
K...maybe this was stated earlier in the conversation...but what's the
deal with the phone? Or was this phone just being carried around by
everyone and ripped apart?
Old Bell Phone + IAXy + 802.11b card +
Hello,
We have an * installation that is causing us fits.
The problems we are seeing:
1) In the middle of a call the call gets dumped and the caller hears a
dial tone.
2) While talking on a call the caller hears nothing for 5 to 10 seconds.
The person on the other end of the call hears
On Apr 5, 2004, at 2:01 PM, Mark Hagler wrote:
Their base rate is $35/mo per peer (single call transit at any given
moment)
and this provides unlimited local and inbound calling. If you are
connecting a PBX and need 1 voice path at any given moment you can
discuss
different pricing
On Mon, Apr 05, 2004 at 11:16:39AM -0500, Bob Klepfer wrote:
Olle E. Johansson wrote:
Richard Airlie wrote:
On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote:
And now leads me to ask... why should my SIP softphones be unable to
register? They are on the same subnet as
On Mon, 2004-04-05 at 15:57, Brian Rathman wrote:
I have the voicemail setup working in that I get the MWI and it emails the
message correctly. When I pressed the MWI button on my SNOM 200, it dials
into the voicemail system and prompts me for a mailbox and password. I know
there is a way to
On Mon, 2004-04-05 at 22:02, Brian Rathman wrote:
I have my server configured to send to send all PSTN traffic to my Cisco
AS5300 gateway via SIP. I use the following line in the extensions.conf file
to accomplish this:
exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],240,T)
And could anybody say the concurrent calls limit for one Asterisk Box? Let's
say it is a Pentium IV 1.6GHz, 256 MB RAM, RedHat 9
thanks,
Pablo Salinas
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I think this is what you are looking
for
Exten = 1000,1,Answer,1Exten =
1000,2,Wait,1Exten = 1000,3,Voicemailmain([EMAIL PROTECTED])
- Original Message -
From:
Mitchell S. Sharp
To: [EMAIL PROTECTED]
Sent: Monday, April 05, 2004 5:27
PM
Subject: Re:
The snom dials into an account caled 'asterisk'
Exten = asterisk,1,Answer,1
Exten = asterisk,2,Wait,1
Exten = asterisk,3,Voicemailmain(${CALLERIDNUM})
- Original Message Follows -
I think this is what you are looking for
Exten = 1000,1,Answer,1
Exten = 1000,2,Wait,1
Exten =
All,
I upgraded to the [*] stable release branch.
When I call into the box (confirmed on 2 installations) the
caller no longer hears the ringing. The CLI confirms that
extensions are being 'rung'.
Whassup?
Willy
Willy Wouters
ypOne Publishing
___
More Info:
And I went back to CVS-03/26/04 and can hear the 'ringing'
again when I call in to the box ...
BTW: This behavior exists on the production system (T1 PRI
interface to PSTN only) and on the Developent system
(FXO/FXS and IAX2 interfaces)
Cheers,
Willy
- Original Message Follows
What do you do when $CALLERIDNUM of the caller isnt the
4-digit extension? I set all of my users Caller ID entries to their 10-digit
phone # so that Caller ID appears correctly when I send their call out the PRI
to the public network. The side effect of this is breaking convenient access
I ran into the same problem. It seems to be fixed in later builds.
-brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, April 05, 2004 5:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Stable Relase
On Apr 5, 2004, at 3:53 PM, Mark Hagler wrote:
What do you do when $CALLERIDNUM of the caller isnt the 4-digit
extension? I set all of my users Caller ID entries to their
10-digit phone # so that Caller ID appears correctly when I send their
call out the PRI to the public network. The side
I use an AGI script I wrote. It's specific to my setup, but you can get
a copy at http://www.fnords.org/~eric/asterisk/downloads/ You'd have to
adapt it to your own needs, of course. Basically it does this: When
called with no options it strips off the first 6 digits of the CallerID
if the
http://www.smh.com.au/articles/2004/04/05/1081017104255.html
SingTel
ready to break into web telephony
April 6, 2004
Singapore
Hola Pablo,
on the box you describe the maximum would be ZERO. You haven't mentioned
what CT hardware you would like to use.
Salud!
On Tue, 2004-04-06 at 07:27, pesb wrote:
And could anybody say the concurrent calls limit for one Asterisk Box? Let's
say it is a Pentium IV 1.6GHz, 256 MB
Title: Message
Try
placing the following in your extensions.conf file:
exten =
1000,1,VoicemailMain(${EXTEN:6})
That
strips the first six number off of the Caller ID leaving the last four digits to
correspond with the voicemailbox. I've got it working on one of my
servers.
Yves Chouinard wrote:
I found some posts regarding this issue dating of September 2003, but no
real answer.
The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I
need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help
migration.
Is there an existing
I had slightly different problems - but the resolutions might help you:
I had a problem with an intermittent loud buzzing on my X100P (heard
when accessing PSTN from my SIP and Zap clients). The problem went
away when I physically moved the card down a PCI slot (further away
from my TDM400P
I'm running into a similar situation. We have 3-digit extensions and a
4-digit DID numbers that get used for for outbound CID. Therefore, no
$CALLERIDNUM direct access to voicemail. Suggestions?
What do you do when $CALLERIDNUM of the caller isnt the 4-digit
extension? I set all of my users
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