What is the dtmf setting on all peers involved in the call?
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: Jonas Kellens jonas.kell...@telenet.be
Sent: Wednesday, January 05, 2011 4:55 PM
To: Asterisk Users Mailing List -
OK, I need to dial a macro from AGI and needs to pass an argument.
Ok, I found an bug report, but it was stated un fixable? really after 5
I found this email in the archive, but no solution other then the dodgy work
I want to know each and every parameter's detail that can be included in
Where can I find this?
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On Thu, Jan 6, 2011 at 11:28 AM, Jeff LaCoursiere j...@sunfone.com wrote:
On Wed, 5 Jan 2011, James Lamanna wrote:
See the following SIP trace.
Where in the world does Asterisk get port 1025 to respond to?
This is asterisk 1.6.x.
I'm sure it would be the NAT
On 01/06/2011 11:34 AM, mgra...@mstvp.com wrote:
We should also be very clear that the Siren codecs are supported on the
Polycom SoundStation conference phones and the VVX-1500 Business Media
Phones. These codecs are not supported in the SoundPoint desk phones.
The SoundPoint series support the
PRICAUSE will give you lots of info on why a call was hungup on. Not
sure if SIP will give you the same.
On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
Does Asterisk, currently using version 1.4, get any more information about
the result of an outbound call made over a
On Jan 6, 2011, at 8:08 PM, Joel Maslak wrote:
On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
Are there reasons to prefer the use of PRI over SIP or SIP over PRI?
I run the PBX for my organization which has about 160 extensions. I
wouldn't even think of
The AstLinux Team is happy to announce the release of AstLinux 0.7.5
with options for both Asterisk 220.127.116.11 and Asterisk 1.4.36. More
information about the release is available on our website:
Direct links to the installation files are
You can user DB for this just make real time configuration of Queue and make
all asterisk server connected to Same DB if more load then use replication
for different server on DB, also So that Quque name should be same for all
server and asterisk can call same agent.
Among all the readers anybody have ever work on Granstream device GXE2504A
which act as ippbx and having GUI to configure and maintain.
We are facing one problem with this device, thsi device reply or adding
codec like ilbc,G.721 which is not supported by our Asterisk server or our
I have tried several settings.
Normally I set it to rfc 2833 on most phone types
(Grandstream/YeaLink/Cisco SPA). Works always.
With Snom you have the option : SIP info : on/off/always
Neither of these settings make any difference...
What setting do you have in your Snom phones ??
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