Hello,
All i have one issue regarding caller id, once i received a call from my SIP
provider it always set caller id with append 1 into
original callerID if a call from USA then there is no problem , but if i
receive a call from other country like INDIA i have also
found callerID part as 191
Hello,
Limiting the call duration with the L-option of the Dial()-command is
working fine, however the announcement is not played.
Dialplan :
exten = _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
exten = _367,n,Dial(SIP/test6,,L(11000,5000,5000))
The call lasts for 11 seconds, but 5 minutes
Take a look at /var/log/asterisk/main or full /if enabled. Perhaps
there is a file not found. try:
exten =
_367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
exten =
_367,n,Set(LIMIT_WARNING_FILE=/path_to_your_audiofiles/file)
# do not add any extension!
On Thu, Nov 11, 2010 at 3:43 AM, Thorsten Göllner t...@ovm-group.com wrote:
Take a look at /var/log/asterisk/main or full /if enabled. Perhaps there is
a file not found. try:
exten = _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
exten = _367,n,Set(LIMIT_WARNING_FILE=/path_to_your_audiofiles/file) #
Found the problem already :
Dial(SIP/test6,,L(11000,5000,5000))
Correct syntax is :
Dial(SIP/test6,,L(11000:5000:5000))
semicolon...
Jonas.
On 11/11/2010 10:43 AM, Thorsten Göllner wrote:
Take a look at /var/log/asterisk/main or full /if enabled. Perhaps
there is a file not found. try:
Hello
How can I run the sip service on asterisk on another port beside 5080?
I mean asterisk will still take sip requests on port:5080 and another custom
port, lets say port:6080
Thanks for any help
--
_
-- Bandwidth
Hi,
I dial on A* from a linphonec to a Playback() extension, then suddenly
the sound stops after a while, without any notice.
I enabled debug both in linphone and A*, and the RTP packets are sent
from A* and received from linphone. It doesn't matter whether I choose
alaw, ulaw, gsm as codec
Paulo Santos wrote:
Hello,
Following my first mail about this issue [1], I think I know now what
the problem is.
When I have both lines being used and a third call comes in, the person
calling doesn't get a busy tone, he gets something like line unavailable.
I've been debugging mISDN
On 11/09/2010 03:20 PM, Gareth Blades wrote:
Jonas Kellens wrote:
On 11/09/2010 02:12 PM, Gareth Blades wrote:
Jonas Kellens wrote:
On 11/08/2010 09:50 PM, Jonas Kellens wrote:
Hello,
SIP DNS SRV records are not working.
My Grandstream uses the SRV records to
Hello
We would like to customize the voicemail menues.
So the intro should not be played if some user has recorded an own greeting
message and we would also like to remove some options from the menue.
Is this all hardcoded or is it somehow possible to redefine the voice menues
and the order
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Benoit Panizzon
Sent: Thursday, November 11, 2010 11:29 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] VoiceMail customizing
Hello
We
I did some more tests, and it's not really a problem with linphone: the
rtp capture shows empty packets sent by Asterisk.
Since the channel which is doing Playback() is in a MeetMe conference, I
tried also to speak on another phone on the same conference: well the
rtp capture shows the stream
The Asterisk Development Team has announced the release of Asterisk
1.4.37. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.37 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk
1.6.2.14. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.14 resolves several issues reported by the
community and would have not been
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Watkins,
Bradley
Sent: Thursday, November 11, 2010 10:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoiceMail
NAT? Firewall?
On Thu, Nov 11, 2010 at 3:21 PM, Marek Soha ma...@snet.sk wrote:
Hi all.
I have an issue with T.38 and re-invites.
Topology:
provider - A (asterisk 1.6) - B (asterisk 1.6) - extension -
- (software fax, gateway whatever).
When between A and B trunk is canreinvite=no
On 11/11/2010 04:21 PM, Marek Soha wrote:
Hi all.
I have an issue with T.38 and re-invites.
Topology:
provider - A (asterisk 1.6) - B (asterisk 1.6) - extension -
- (software fax, gateway whatever).
When between A and B trunk is canreinvite=no everything is working
smooth. When I
Does it matter?
Phones are working correctly...I tried also portforwarding.
So corrected topology:
provider - A (asterisk 1.6) - B (asterisk 1.6) - extension -
- NAT/FIREWALL - (software fax, gateway whatever).
Software fax ends with DIS sent, 9600Bbps
Joel, dňa 11. novembra 2010 ste napísali:
On 11/11/2010 04:48 PM, Marek Soha wrote:
Uf...
you are perfectly clean about that confusion...
Only thing I want to do, is to route stream out of local asterisk - to
connect final extension directly to sender - provider.
So what I can do if I need i.e:
1) canreinvite=yes AND send T38
On Tue, 9 Nov 2010, Daniel Tryba wrote:
I am curious about the tool dahdi_maint... what do the various
acronyms stand for?
Yea there seemed to be a bit of confusion here as well so I patched
trunk with some more descriptive error counter labels :O)
FEC : 0:
Framing Errors
CEC : 0:
CRC
On Thu, 11 Nov 2010, Russ Meyerriecks wrote:
On Tue, 9 Nov 2010, Daniel Tryba wrote:
I am curious about the tool dahdi_maint... what do the various
acronyms stand for?
Yea there seemed to be a bit of confusion here as well so I patched
trunk with some more descriptive error counter labels
On 11/11/10 5:44 PM, Jeff LaCoursiere wrote:
On Thu, 11 Nov 2010, Russ Meyerriecks wrote:
On Tue, 9 Nov 2010, Daniel Tryba wrote:
I am curious about the tool dahdi_maint... what do the various
acronyms stand for?
Yea there seemed to be a bit of confusion here as well so I patched
trunk
On Thu, 2010-11-11 at 18:28 -0600, Russ Meyerriecks wrote:
On 11/11/10 5:44 PM, Jeff LaCoursiere wrote:
On Thu, 11 Nov 2010, Russ Meyerriecks wrote:
On Tue, 9 Nov 2010, Daniel Tryba wrote:
I am curious about the tool dahdi_maint... what do the various
acronyms stand for?
Yea
On 11/11/10 7:23 PM, Carlos Chavez wrote:
I seem to be having the same problem with a new server. I am using a
TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2.13 on
a Dell server. All calls to the outside have bad voice quality (echo
and distortion). Internal calls
You try install debian in your sparc platform ?
On Thu, Nov 11, 2010 at 8:52 PM, RR ranjt...@gmail.com wrote:
Hello Group,
I have been going through all the chit-chat about TTS and the various
engines available to integrate with Asterisk incl. flite/festival, espeak,
Nuance etc but I am
No, I want to use Solaris 10 on the Sparc platform. I've read a lot of
reports and tests/benchmarks conducted that sow Solaris 10 actually
performing better than all other Linux based Distros...not sure if that's
been the experience of others in the group.
I really want to know if someone has a
I use Nuance, festival, Ibm tts and Loquendo.
Now in your case, i suggest use tts on the recommend tts
environment. Solaris is not standart system for tts products. Then you
can plug tts system into asterisk platform.
I use Debian for sparc and work excelent!! don't discard this option
may be
On Thu, 11 Nov 2010 20:08:09 -0600, Russ Meyerriecks wrote
On 11/11/10 7:23 PM, Carlos Chavez wrote:
I seem to be having the same problem with a new server. I am using a
TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2.13 on
a Dell server. All calls to the outside
Hi Luis,
Thanks for your comments. How / Why are you using that many TTS products? Do
you have a preference of one over the other?
Also, do you have any documentation / install/configuration notes that you
might be willing to share re: your experience with Debian on Sparc and the
TTS
Well,
I use many tts products because i work with diferents telphone
systems. Now for asterisk the best way for free is Festival and noon
free is Loquendo.
I'm not have notes to install debian on Sparc, i just only use debian
readme :-) It's too easy, debian work for you :D
Just download
asterisk by default listen on port 5060.You simply need open the file
/etc/asterisk/sip.conf and change these.
udpbindaddr=0.0.0.0:6080tcpbindaddr=0.0.0.0:6080save the file and open
asterisk console and execute sip reload.
Muhammad Faheem
--- On Fri, 11/12/10, Baha @ SH i...@saudihome.com
Sure, no worries. Will try that. What about advice on TTS setup. Would you
have any notes on how best to setup high-volume TTS environment, like maybe
a cluster of TTS servers and how Asterisk talks to those? Recommendations on
how to set that up? I'm thinking about trying Festival/FLite and maybe
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