Re: [asterisk-users] Asterisk as register server through OpenSIPS

2012-01-10 Thread Olle E. Johansson
9 jan 2012 kl. 09:02 skrev Ronald Cepres: Hi all, I've been trying to register a SIP user agent to an Asterisk server using OpenSIPS as SIP router. The functionality is working fine. However, Asterisk uses the IP address of the OpenSIPS server as the peer IP address. How can I use the

Re: [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?

2012-01-10 Thread Johann Steinwendtner
Its the sending complete IE. I'm using EuroISDN and I also use overlap signalling on this interface. Regards Hans On 2012-01-10 01:12, C F wrote: Exactly which IE message are you trying to push manually? you shouldn't have to do that, it should be done in the configs for you. On Mon, Jan 9,

Re: [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0)

2012-01-10 Thread Olivier
Hi, 1. This patch didn't correct the issue but I'm far from certain that I correctly applied the patch. 2. I took the Hardware Echocan module off my board and it seems to correct the issue. I'll dig deeper to check if I correctly applied the patch and both report here and in DAHLIN-275 ticket.

Re: [asterisk-users] Connecting to an Old Phone System

2012-01-10 Thread Paul Hayes
On 06/01/12 13:14, Dan Journo wrote: Is there such a thing as an ISDN30e PCI card which can be used with a copy of Asterisk, that can act like a voip gateway between the old phone system, and our asterisk box? Yes Digium sell 2 port PRI cards that support E1. TE200 series. I use them like

Re: [asterisk-users] Change port from 5060 on Snom phone

2012-01-10 Thread Paul Hayes
On 06/01/12 16:17, Ishfaq Malik wrote: Hi Does anyone know how to change the target port on a Snom phone. I have tried adding :new port number to the end of the registrar but this doesn't work. It should do. Try putting registrarip:port into Outbound Proxy and leave the Registrar box just

Re: [asterisk-users] Change port from 5060 on Snom phone

2012-01-10 Thread José Pablo Méndez Soto
Can you email me off list (since this isn't really Asterisk related and a snom support issue, which I can help with) with some details and ideally a SIP trace? cheers, Paul. Closing this question with a final message including the [SOLVED] phrase will definitely help the community I

Re: [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0) [SOLVED]

2012-01-10 Thread Olivier
2012/1/10, Olivier oza_4...@yahoo.fr: Hi, 1. This patch didn't correct the issue but I'm far from certain that I correctly applied the patch. I was right to suspect I was wrong : now, after correctly applying the DAHLIN-275 patch, it's working OK (with the EchoCan module plugged-in). Thanks

[asterisk-users] Hang up phone after declined attended transfer

2012-01-10 Thread Carlos Alvarez
We have a customer who has asked us to change this behavior, but I haven't been able to find a way to do it. Server is Asterisk 1.6 and the phones are SPA 303 and 504. Receptionist gets an outside call, starts an attended transfer The office person being called answers by pressing the speaker

Re: [asterisk-users] Hang up phone after declined attended transfer

2012-01-10 Thread Danny Nicholas
You could use a parking lot instead of attended transfer? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Tuesday, January 10, 2012 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Re: [asterisk-users] Hang up phone after declined attended transfer

2012-01-10 Thread Carlos Alvarez
On Tue, Jan 10, 2012 at 10:24 AM, Danny Nicholas da...@debsinc.com wrote: You could use a parking lot instead of attended transfer? Since that actually is more work than just training all the users to manually hang up, I have a feeling the customer won't be enthusiastic about it. But thanks

Re: [asterisk-users] Hang up phone after declined attended transfer

2012-01-10 Thread Ryan Wagoner
On Tue, Jan 10, 2012 at 12:02 PM, Carlos Alvarez car...@televolve.comwrote: We have a customer who has asked us to change this behavior, but I haven't been able to find a way to do it. Server is Asterisk 1.6 and the phones are SPA 303 and 504. Receptionist gets an outside call, starts an

Re: [asterisk-users] Hang up phone after declined attended transfer

2012-01-10 Thread Ryan Wagoner
On Tue, Jan 10, 2012 at 1:57 PM, Ryan Wagoner rswago...@gmail.com wrote: On Tue, Jan 10, 2012 at 12:02 PM, Carlos Alvarez car...@televolve.comwrote: We have a customer who has asked us to change this behavior, but I haven't been able to find a way to do it. Server is Asterisk 1.6 and the

Re: [asterisk-users] Hang up phone after declined attended transfer

2012-01-10 Thread Carlos Alvarez
On Tue, Jan 10, 2012 at 12:01 PM, Ryan Wagoner rswago...@gmail.com wrote: I did a quick search and found the setting. Go to the Regional tab and find the Reorder Delay. Change that to 255, which will disable the order tone and cause the phone to hangup. Thanks, that did it! Now to

[asterisk-users] Linux Stun Server

2012-01-10 Thread Bryant Zimmerman
I have been running the windows vovida stun server for some time and has worked without issue, but I really want to run a linux stun server and get away from the windows based one. Anyone have an idea of a good replacment that can be compled on opensuse? Thanks Bryant --

Re: [asterisk-users] Linux Stun Server

2012-01-10 Thread Danny Nicholas
Why don't you just use vovida-linux from sourceforge? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Tuesday, January 10, 2012 3:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Linux Stun

Re: [asterisk-users] Best non polycom SIP conference room phone

2012-01-10 Thread Jamie A. Stapleton
Snom is an OEM of the Konftel. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of brya...@zktech.com Sent: Sunday, January 08, 2012 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Re: [asterisk-users] Best non polycom SIP conference room phone

2012-01-10 Thread Bryant Zimmerman
So are the Konftel conference room phones any good? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Jamie A. Stapleton jstaple...@computer-business.com Sent: Tuesday, January 10, 2012 5:25 PM To: Asterisk Users Mailing List -

[asterisk-users] Odd DTMF problem when receiving calls

2012-01-10 Thread Christopher David Howie
Hello, I've been tinkering with Asterisk today for the fun of it, trying to set up my own domain. I've got pretty much everything working, including a DID number that connects to my extension. However, I'm having a problem receiving calls from a particular peer, specifically my office's PBX,

Re: [asterisk-users] Odd DTMF problem when receiving calls

2012-01-10 Thread David Backeberg
On Tue, Jan 10, 2012 at 6:00 PM, Christopher David Howie m...@chrishowie.com wrote: I've been up and down this issue for a few hours and I cannot for the life of me determine why simply defining a peer causes Asterisk to offer telephone-event.  I have tried specifying dtmfmode=rfc2833 or

[asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??

2012-01-10 Thread shalu dhamija
Hello, I am trying to run load on asterisk server(version 1.8.7.1) for the voicemail() application using SIPp tool. I am just running sipp at call rate of 1 cps with the following command: ./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s

Re: [asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??

2012-01-10 Thread virendra bhati
Hi Shalu, How you are invoking call in dialplan. it's completely depends on that. And error show that no voice is there for store in voicemail . On Wed, Jan 11, 2012 at 10:05 AM, shalu dhamija shalu.dham...@rancoretech.com wrote: Hello, I am trying to run load on asterisk server(version

[asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?

2012-01-10 Thread Olivier
Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting

Re: [asterisk-users] Anyone have a reliable T.38 Solution

2012-01-10 Thread Olivier
2012/1/5, Kevin P. Fleming kpflem...@digium.com: On 01/04/2012 12:25 AM, Matt Darnell wrote: Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI-- Asterisk-- T.38-- ATA-- Fax Does version 1.8 with the Digium fax driver have this capability?

[asterisk-users] Problems faced in load testing of asterisk

2012-01-10 Thread shalu dhamija
Hello, I am trying to run load on asterisk server(version 1.8.7.1) through SIPp tool for the voicemail() application. But I am facing a lot of problems. I tried running 1000 calls from SIPp for 100 subscribers (10 messages for each subscriber). I am using odbc storage for the messages.