Re: [asterisk-users] [IAX] Everyone is busy/congested at this time (1:0/0/1)

2011-04-23 Thread John Alexis
Hi, Sorry to insist, but I still not have any solution. Does anybody have an idea ? Thanks! 2011/4/20 John Alexis kasteris...@gmail.com Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine

[asterisk-users] Warm Transfer in Asterisk

2011-04-23 Thread Jeff Johnson
Is there a way do what is sometimes called a 3rd party transfer in Asterisk. That is; Call A comes in and is answered B. B then places A on hold and calls C. After C answers, BC chat for a moment, then B brings A on line. After making intro's B then drops off call. Thanks, Jeff

Re: [asterisk-users] Warm Transfer in Asterisk

2011-04-23 Thread Ryan Wagoner
On Sat, Apr 23, 2011 at 8:56 AM, Jeff Johnson jjohn...@neturallyspeaking.com wrote: Is there a way do what is sometimes called a 3rd party transfer in Asterisk.  That is; Call A comes in and is answered B.  B then places A on hold and calls C.  After C answers, BC chat for a moment, then B

Re: [asterisk-users] Cannot call to my server with SIP

2011-04-23 Thread Paul van der Vlis
Op 22-04-11 22:58, Steve Edwards schreef: Op 22-04-11 18:13, Eric Wieling schreef: sip set debug on should help On Fri, 22 Apr 2011, Paul van der Vlis wrote: I've tried it, but no, nothing... Sounds like you have very basic network issues. Can this host ping your SIP endpoint?

Re: [asterisk-users] Warm Transfer in Asterisk

2011-04-23 Thread Jeff Johnson
I'll give it a whirl, Thanks, Jeff -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: Saturday, April 23, 2011 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Re: [asterisk-users] Cannot call to my server with SIP

2011-04-23 Thread Paul van der Vlis
Op 22-04-11 23:49, Jamie A. Stapleton schreef: I can see your server just fine... -bash-3.2# ./svmap.py xen8.vandervlis.nl | SIP Device | User Agent | Fingerprint | -- | 91.198.178.28:5060 |

[asterisk-users] DTMF not being sent ( RFC2833 )

2011-04-23 Thread David
Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with

[asterisk-users] call files

2011-04-23 Thread Tiago Geada
Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450@ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTime: 60 Account: Outbound210332450 Context:

Re: [asterisk-users] call files

2011-04-23 Thread Tiago Geada
Hi, Using DumpChan(); Seems that Channel (where the call goes first) is a sub-channel of Context/Extension (where the call goes on CONNECT) ?? first I have: Dumping Info For Channel: Local/210332450@ZonNew-Outbound-66c7;2: Then after: Dumping Info For Channel:

Re: [asterisk-users] call files

2011-04-23 Thread Sherwood McGowan
On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada tiago.ge...@gmail.com wrote: Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450@ZonNew-Outbound CallerID:

[asterisk-users] ARA table definitions (1.8.*)

2011-04-23 Thread Jason Rogers
Where would one find, or better yet determine from code, all of the table definitions for ARA dynamic families? There seems to be some bits and pieces in various places around the internet, ie. voip-info, the definitive guide, ect. but nothing complete or definitive. I have wondered about this

Re: [asterisk-users] ARA table definitions (1.8.*)

2011-04-23 Thread Hans Witvliet
On Sat, 2011-04-23 at 10:52 -0700, Jason Rogers wrote: Where would one find, or better yet determine from code, all of the table definitions for ARA dynamic families? There seems to be some bits and pieces in various places around the internet, ie. voip-info, the definitive guide, ect. but

Re: [asterisk-users] Huawei K3765 + Internet + SMS + Telephone

2011-04-23 Thread Michelle Konzack
Hello asterisk asterisk, Am 2011-04-23 06:24:36, hacktest Du folgendes herunter: Look at this wiki for help. http://wiki.e1550.mobi/doku.php For asterisk, you can use your USB stick for voice/SMS but not internet at the same time. A separate internet connect is required per my

Re: [asterisk-users] Cannot call to my server with SIP

2011-04-23 Thread Eric Wieling
If you don't see the call coming in when you have sip debug enabled, then the call is not making it to your server. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul van der Vlis Sent: Friday, April 22,

Re: [asterisk-users] Nat=yes

2011-04-23 Thread Pezhman Lali
check this http://www.voip-info.org/wiki/view/Asterisk+sip+nat On Thu, Apr 21, 2011 at 2:12 PM, Alexandru Oniciuc alexandru.onic...@trivenet.it wrote: Dear * users, in your opinion, when using a * as a public server, is good practice enabling nat=yes in sip.conf for all the peers? Can

Re: [asterisk-users] IAX2 codec selection and video

2011-04-23 Thread Pezhman Lali
check this url, let me know if any problem http://www.voip-info.org/wiki/view/Asterisk+video http://www.voip-info.org/wiki/view/Asterisk+video http://www.voip-info.org/wiki/view/Asterisk+videobest On Thu, Apr 21, 2011 at 9:00 PM, Steve Davies davies...@gmail.com wrote: Hi, Can anyone let