Re: [asterisk-users] DTMF-troubles with Snom

2011-01-08 Thread Bryant Zimmerman
Jonas What is the dtmf setting on all peers involved in the call? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Jonas Kellens Sent: Wednesday, January 05, 2011 4:55 PM To: Asterisk Users Mailing List -

[asterisk-users] AGI-Macro w/Agruments

2011-01-08 Thread William Stillwell
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 years? I found this email in the archive, but no solution other then the dodgy work around?

[asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?

2011-01-08 Thread Bruce B
Hi Everyone, I want to know each and every parameter's detail that can be included in the read= write= in manager.conf Where can I find this? Thanks -- _ -- Bandwidth and Colocation Provided by --

Re: [asterisk-users] Asterisk replying to wrong port for NOTIFY messages

2011-01-08 Thread James Lamanna
Hi Jeff, On Thu, Jan 6, 2011 at 11:28 AM, Jeff LaCoursiere 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. Hi James, I'm sure it would be the NAT

Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?

2011-01-08 Thread Kevin P. Fleming
On 01/06/2011 11:34 AM, 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

Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

2011-01-08 Thread C F
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 wrote: Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a

Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

2011-01-08 Thread Tom Rymes
On Jan 6, 2011, at 8:08 PM, Joel Maslak wrote: On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson wrote: Are there reasons to prefer the use of PRI over SIP or SIP over PRI? [snip] I run the PBX for my organization which has about 160 extensions. I wouldn't even think of

[asterisk-users] AstLinux 0.7.5 released

2011-01-08 Thread Darrick Hartman (lists)
The AstLinux Team is happy to announce the release of AstLinux 0.7.5 with options for both Asterisk and Asterisk 1.4.36. More information about the release is available on our website: Direct links to the installation files are

Re: [asterisk-users] Call queues on load-balanced asterisks

2011-01-08 Thread DHAVAL INDRODIYA
Hello Pan, 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. you didnot

[asterisk-users] Grandstream GXE2504A codec disable option

2011-01-08 Thread amit salunkhe
Dear All 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 SBC.

Re: [asterisk-users] DTMF-troubles with Snom

2011-01-08 Thread Jonas Kellens
Hello, 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 ??