Ciao Eric,
If you had a PRI (not just a T-1) AND your telco permits you to set
it.
Is there any hope to change the caller-id on a BRI line?
Thanks,
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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hi can we do autocreatepeer in iax.conf?
thanks
kAMRAN
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On Sat, 2006-08-05 at 00:09 -0700, Kamran Ahmad wrote:
hi can we do autocreatepeer in iax.conf?
No, that option is not available for iax.conf.
--
Russell Bryant
Software Developer
Digium, Inc.
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Andrea Spadaccini wrote:
Is there any hope to change the caller-id on a BRI line?
I can change my Caller ID on my BRI lines to anything within my DID range.
Hope that helps,
Avi
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Using svn trunk, I am trying to record a call coming in from a zap line.
This works:
A) Zap-StartRecording-DialSip-xferToSIP2-Talk-Hangup
The entire call is recorded until the hangup
However, if you use a local channel instead:
B) Zap-StartRecording-DialLocal-DialSip-xferToSIP2-Talk-Hangup
Hi,does anybody used cisco 2600 as * gateway with E1?Thanks-- .:FaberK:.
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asterisk-users mailing list
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Add /n to you Local dial string, i.e.:
Dial(Local/1234/n)
From http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels :
Adding /n at the end of the string will make the Local channel not
do a native transfer (the n stands for no release) upon the remote
end answering the line.
Hi.
Tks.
What it is TAPI license and how much we have to pay for that?
Cris.
From: (AstATN) [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MWI from
Hello,
I have a fax server with an AVM Fritzcard that is connected to port number 4
of an EICON DIVA Server 4 BRI. As you can see from the following debug
messages, asterisk is accepting the incoming fax call on ISDN4 and forwards
it to port number 3 (ISDN3).
But at the end the call is
Thanks for the reply, but as I mentioned in the original email (option
C), the /n causes problems with warnings on the console.
Julian (L. S.) :)
Julian J. M. wrote:
Add /n to you Local dial string, i.e.:
Dial(Local/1234/n)
From
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
After following the instructions for getting GoogleTalk to talk with
Asterisk using Matt O'Gorman's code and Farruk Ahmed's howTo (
http://www.sineapps.com/news.php?rssid=1407 ) I have been using the
Jabber and Jingle additions in the SVN Trunk
IAX Provider-- Asterisk/NAT Firewall-- IAXy (192.168.0.x)
When a call comes into the Asterisk box, it then rings the IAXy.
When the IAXy is answered, I get..
-- IAX2/homeiaxy-6 answered IAX2/teliax-4
-- Channel 'IAX2/teliax-4' unable to transfer
-- Channel 'IAX2/homeiaxy-6' unable to transfer
Hi All,
Does anyone here have Japanese version of the asterisk sound files?
TIA
Regards
Nhadie
Message sent using UebiMiau 2.7
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Andrea Spadaccini wrote:
Ciao Eric,
If you had a PRI (not just a T-1) AND your telco permits you to set
it.
Is there any hope to change the caller-id on a BRI line?
Sorry, I was being USA-centric. It's a bad habit to get into.
As I understand it, if you have a BRI and your telco allows
Nhadie wrote:
Does anyone here have Japanese version of the asterisk sound files?
http://www.google.com/search?q=japanese+sound+files+site%3Avoip-info.org
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asterisk-users mailing list
To
Hi
I experimented with using the native MOH player with Asterisk 1.2.x
instead of using mpg123. However I discovered that with a queue playing
MOH to 20 waiting callers I was getting a load average 1.00+ using the
Asterisk native compared to 0.08 using mpg123.
Is this normal? If so what is the
Andrea Spadaccini wrote:
Is there any hope to change the caller-id on a BRI line?
I guess you can do it within the range assigned to you. If you have 2
numbers, you can choose between these two numbers. Not tested, as I have
only 1 number here (and still fighting with the zaphfc: empty HDLC
voiplist wrote:
Is there a command to check the call duration of an active call in
the CLI?
show channels verbose
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Hermann Wecke wrote:
voiplist wrote:
Is there a command to check the call duration of an active call in
the CLI?
show channels verbose
show channel channel_id_originating_channel
shows among other things
Elapsed Time: 0h2m47s
Regards
Jon
--
Jon Farmer
Telford, Shropshire, UK
Hi, thanks to the original poster, I redid all the cabling and immediately
got the span to go OK between asterisk and the siemens legacy PBX. Only
problem now is working out how to handle the calls from the siemens
Worth pointing out at this stage I have no access to the siemens
configuration,
James Arscott wrote:
Hi, thanks to the original poster, I redid all the cabling and immediately
got the span to go OK between asterisk and the siemens legacy PBX. Only
problem now is working out how to handle the calls from the siemens
Worth pointing out at this stage I have no access to
do you have notransfer=no in iax.conf iaxy entry?
On 8/5/06, Jeff Iddings [EMAIL PROTECTED] wrote:
IAX Provider-- Asterisk/NAT Firewall-- IAXy (192.168.0.x)
When a call comes into the Asterisk box, it then rings the IAXy.
When the IAXy is answered, I get..
-- IAX2/homeiaxy-6 answered
Title: Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Hi Jon, thanks for the reply, good to know I am on the right track!
I have actually already tried your suggestion, but using :
_. To try and match what the siemens was sending, but this did not work, I know this is not advisable in
Negative.
Moises Silva wrote:
do you have notransfer=no in iax.conf iaxy entry?
On 8/5/06, Jeff Iddings [EMAIL PROTECTED] wrote:
IAX Provider-- Asterisk/NAT Firewall-- IAXy (192.168.0.x)
When a call comes into the Asterisk box, it then rings the IAXy.
When the IAXy is answered, I get..
- Original Message -
From: Jeff Iddings
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion [mailto:[EMAIL PROTECTED]
Sent: Sat, 05 Aug 2006
17:15:58 -0300
Subject: Re: [asterisk-users] IAXy behind NAT, unable to
transfer?
On 8/5/06, Jeff Iddings [EMAIL
Joshua Colp wrote:
- Original Message -
From: Jeff Iddings
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion [mailto:[EMAIL PROTECTED]
Sent: Sat, 05 Aug 2006
17:15:58 -0300
Subject: Re: [asterisk-users] IAXy behind NAT, unable to
transfer?
On
- Original Message -
From: Jeff Iddings
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion [mailto:[EMAIL PROTECTED]
Sent: Sat, 05 Aug 2006
17:36:00 -0300
Subject: Re: [asterisk-users] IAXy behind NAT, unable to
transfer?
Fair enough, but my IAXy
Hi!
I have two extensions (25 and 26, and so two phones) for one person in
an office.
I can dial 25 or 26 and always both extensions are ringing. Thats okay!
exten = 25,1,Dial(Sip/25Sip/26)
exten = 26,1,Dial(Sip/25Sip/26)
The problem with this solution is, if the person is talking on one phone
Thomas Artner skrev:
Hi!
I have two extensions (25 and 26, and so two phones) for one person in
an office.
I can dial 25 or 26 and always both extensions are ringing. Thats okay!
exten = 25,1,Dial(Sip/25Sip/26)
exten = 26,1,Dial(Sip/25Sip/26)
The problem with this solution is, if the person
Marcus Carlson wrote:
Thomas Artner skrev:
Hi!
I have two extensions (25 and 26, and so two phones) for one person in
an office.
I can dial 25 or 26 and always both extensions are ringing. Thats okay!
exten = 25,1,Dial(Sip/25Sip/26)
exten = 26,1,Dial(Sip/25Sip/26)
The problem with this
I'm new to Asterisk and am trying to write an AGI script in perl and need
some pointers. The script simply plays a few gsm files in succession
before doing a database insert (using perl's DBI in a sub). In a nutshell,
it looks like this:
print EXEC Playback foo1\n;
print EXEC Playback foo2\n;
i am install server asterisk with 266Mhz , disk scsi and 380 Megas of ram.i am install into system the licence the codec g729 and the call it happens through the asterisk??the packets nessesarity through server asterisk ?
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This might be what you're seeking;
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail
If the phone rings, then the channel IS available. The solution is to
disable call waiting on the SIP device.
The s option needs to be used:
s - Consider the channel unavailable if
Anybody have a recent copy of the Admin Guide (not the
user guide) for the SPA3000/3102? The only one I was
able to find was a terribly written two year old one
on the Sipura site[1] and Linksys says you have to be
a Service Provider to get one from them.
There is no rtcachefriends in the standard set up
of real time. shoul I add it as a row to the DB or should I have it in sip.conf
under the general section ?
Dovid
- Original Message -
From:
William
Piper
To: Asterisk Users Mailing List -
Non-Commercial Discussion
On Sat, 2006-08-05 at 20:43 -0400, Roy Kidder wrote:
Is there some way I can better control the execution of playbacks so that
they take place as I expect them to?
Yes, your script needs to read a line of input from stdin to wait for
Asterisk to send back the result code indicating that the
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