similar to: Background noise

Hi ho, is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes (1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1 being created upon SIP client registration, "show dialplan xxx" reveals no change. And yes, I have also read and checked bug 7144; if I go down that route and no

Hi All, Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 ?Payment Required? from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the

Hi, looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. -- Thanks, Phil

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi there, is gizmo the first user of the Digium Skype solution, or do they use a different approach/product - any clue? http://www.gizmo5.com/pc/opensky/ Philipp

Hi, I'm tying to delay the HungUp. I tried this way: exten => h,1,NoOp(Start) exten => h,n,Wait(5) exten => h,n,NoOp(End) exten => h,n,Hangup() but it doesn't work, Any idea? Thanks in advance.

Ok, here's the challenge: I would like to be able to find, match - and then react - upon prompts that are presented by the outbound/remote side of a call. Think mobile phone and "This user is temporarily unavailable". Collecting a limited number of known prompt snippets should not be a problem, but how would you then detect their presence in a longer recording (or live audio

I didn't know if it would work or not, but I figured I'd try slow-speed half-duplex TDD over GSM & Vonage. I called a AGI script I have that speaks to TTYs, by calling from Vonage to one of my Voicepulse lines. I don't control the Vonage codec, so I have no idea what it uses, but I am using GSM for the Voicepulse line. Everything worked fine - echo canceling didn't cause any

I just moved my system over to a new server with * 0.7.1. The old machine was using a cvs from August/Sep timeframe. On the new machine I did an make samples but then ovewrote with tar files of the production configs in the /etc/asterisk /var/spool/asterisk /var/lib/asterisk folders. Now the system seems to be working fine but only records blank audio in the voicemail files. Same thing with

Hi, I seem to have a problem with chanisavail and the call limits on sip phones(incoming and outgoing) The problem seems to be that chanisavail when trying create to create channels and hanging them up afterwards screw up the current usage limit on the phones. Example with chanisavail: Phone A calls voicemail (usage now 1) Phone B tries to call Phone A and uses ChanIsAvail in the dialplan.

Hi, For various reasons a customer of mine is moving from a SER-based to an Asterisk-based installation, mostly because of problems with SIP devices behind NAT trying to reach each other and because it's easier to do accounting when all calls go through Asterisk (canreinvite=no is the idea). The database-based SIP registration mechanism of Asterisk seems to have one shortcoming - it

Hi, I am registering my Asterisk boxes to a SIP provider for outgoing calls. My "outgoing" dialplan context tries to dial out in sequence, starting with the SIP provider then ISDN lines and finally analog lines. So the idea is that if the SIP trunk fails then all calls are dialed out via ISDN and analog. I noticed however that if I switch my DSL connection off (ie. no internet access

Anyone got any subjective (!) views on the merits of these two ranges , using asterisk 1.4 ? I need to supply approx 30 handsets to a new client, with the senior managers (6) having some slightly more "managerial" phones than the base phones which will be used for one line only. TIA Julian -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am trying to find remote call forwarding feature in asterisk. I don't know is it possible or any one had already done it. SBC (local Telco) provide such feature. I can call into my voicemail number, and set the remote-call-forward to my cell or another number. It is like person can remotely manage to set the call-forward or DND to his/her extension. Can this be doable in asterisk?

Greetings, Sounds like a simple thing to do, but I was not able to do it on these particular phones. Snom wiki was not helpful. My client wants to keep his phones pointed to UTC time, no DST, no change in timezone, i.e. to stay at 0 hours difference. The phones are provisioned from a tftp server. If I remove 'dst' value from the provisioning file, on bootup phones force users to pickup

Hi, 1. Do you have any experience with receiving incoming SMS on an analog or ISDN landline ? How can then you differentiate an SMS call from a voice call ?

Hi Guys, i have a small issue but bothering me, after restarting asterisk (version 1.4 running on centos) i have the following message that comes repeatedly when i am connected to the CLI: -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected does any one know how to stop this or if it's a sign of a

Good afternoon list. I'm experiencing a problem with my SIP channel's. When I have an external connection for one of my SIP carrier's, I can listen to the client and the client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. In the console of Asterisk generates the

Hello: Here is my sip and extentions configuration and the log of x-lite, because i don`t can call inside my LAN with asterisk PBX 1.2 and i don`t have NAT. i hope you can help me. SIP.conf [default] include=>anexos include=>anexos1 include=>anexos2 [anexos] exten=> 100,1,Dial(SIP/100,0) exten=> 100,2,Hangup [anexos1] exten=> 101,1,Dial(SIP/101,0) exten=> 101,2,Hangup