[asterisk-users] call between snom 300 and aastra 6731i

hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and

Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]

I use a Perl script that monitors AMI events. It also checks the state of all queues and members and generates some basic HTML pages for monitoring the queues. It's not perfect, nor would I call it pretty, but it gets the job done. If you are interested, I can send it to you. Dale On Wed, Mar

Re: [asterisk-users] Anonymous SIP calls

You have to consider whether you really want anonymous calls, or you just want to enable SIP calls from trusted companies/partners. The latter means setting up routes to these companies and (ideally) registration between peers. If you really want anonymous calls, then you will have to setup

[asterisk-users] Gateway Eurotech

Hi, I know there are people with much experience in asterisk, and I want to ask if anyone had experiance with this gw http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/ I'm having trouble getting connect with asterisk anyone has any production? regardss

Re: [asterisk-users] Auto Answer

2015-03-23 11:08 GMT-06:00 ricky gutierrez xserverli...@gmail.com: Hi , I'm having some problems with functions enable auto answer in some Grandstream GXP 1405 , I have enabled this feature in the snom 821 phone and work gr8 , in the gandstream not work, I enable the function on the phone

Re: [asterisk-users] CDR dst value null after attended transfer

On Thu, Mar 26, 2015 at 10:24 AM, Vinicius Fontes vinic...@aittelecom.com.br wrote: I'm having an issue with CDR. Basically, I expect to have all legs of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an

[asterisk-users] Anonymous SIP calls

We have a FreePBX-12 / Asterisk-12 setup that supports about 24 extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence. The server host is a dedicated atom(tm) box using the FreePBX distro (CentOS-6.x) and is up-to-date.

[asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown

I found an issue with how PJSIP handles a typo in the Dial application. If the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//...), the Dial applications fails (obviously), but it also kills the server. I put some code in my pbx_config to check for that string and not let the

Re: [asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown

On Thu, Mar 26, 2015 at 9:28 AM, Trey Hilyard kct...@gmail.com wrote: I found an issue with how PJSIP handles a typo in the Dial application. If the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//...), the Dial applications fails (obviously), but it also kills the server. I

[asterisk-users] CDR dst value null after attended transfer

I'm having an issue with CDR. Basically, I expect to have all legs of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm