Re: [asterisk-users] Make outgoing calls through BroadWorks/BroadSoft SIP gateway from Asterisk

2012-08-18 Thread James Stocks
On 18 Aug 2012, at 09:45, James Stocks wrote: > I've been given a SIP hard phone pre-configured to work with another party's > BroadWorks system. I want to use my Asterisk system to connect to this SIP > service rather than the handset I've been given. I have extract

[asterisk-users] Make outgoing calls through BroadWorks/BroadSoft SIP gateway from Asterisk

2012-08-18 Thread James Stocks
I've been given a SIP hard phone pre-configured to work with another party's BroadWorks system. I want to use my Asterisk system to connect to this SIP service rather than the handset I've been given. I have extracted the authentication details from the phone and have successfully registered A

[asterisk-users] chan_mobile with Nokia 6021 - incoming SMS causes call to drop

2012-04-30 Thread James Stocks
Hello, I'm Using Asterisk 1.8.11.0 on Debian Squeeze. I was experiencing problems with ${SMSSRC} being blank, so I applied this patch: https://issues.asterisk.org/jira/secure/attachment/42026/sms-sender-fix.diff but otherwise everything is standard. As the subject says, if I am making a call

[asterisk-users] Park() ignores 'r' option which should disable music on hold in favour of ringing tone

2012-02-16 Thread James Stocks
When I receive a call, I want to automatically park it from the dialplan so that I can retrieve it later. However, I don't want callers to be aware that they are being parked, so I want to play a ringing tone to the caller. Park() is supposed to be able to do this: Park([timeout][,return_c

Re: [asterisk-users] compile issues.

2009-12-25 Thread James Stocks
On 25 Dec 2009, at 11:36, --[ UxBoD ]-- wrote: > - "Aditya Kumar" wrote: > > No. I am not installing as ROOT. > I dont want to install in ROOT? if so what all should I do> You need to become root in order to build and install Asterisk. This doesn't mean that you have to run Asterisk as r

Re: [asterisk-users] sendmail

2009-12-20 Thread James Stocks
On 19 Dec 2009, at 16:20, Thomas Perron wrote: > Anyone have a cookbook on configuring sendmail to work with Asterisk? > Or,a few config examples. Postfix is a drop-in replacement for sendmail. I find it to be far, far simpler to administer. Take a look at http://www.postfix.org/documentation.

Re: [asterisk-users] Example to handle incoming calls without callerid at home?

2009-12-06 Thread James Stocks
On 6 Dec 2009, at 08:56, Remco Barendse wrote: > I am using asterisk 1.6 at home and would like to send incoming calls > without caller id immediately to voicemail (i don't want to use the > privacy manager where people have to enter a number). > > The config examples i found are all for the pr

Re: [asterisk-users] call log, call detail

2009-11-15 Thread James Stocks
On 15 Nov 2009, at 11:39, wrote: > hi friends, > > i had installed postgres database for call log,call detail. it has restarted > succesfully but when i check tcp connection i dont get any welcome message > by psql. > > [r...@localhost ~]# # psql -h 127.0.0.1 -U asterisk password > [r...

Re: [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image

2009-10-11 Thread James Stocks
On 3 Oct 2009, at 20:38, James Stocks wrote: > On 3 Oct 2009, at 16:37, Jonathan Thurman wrote: > >> On Sat, Oct 3, 2009 at 6:17 AM, James Stocks >> wrote: >>> Hi everyone, >>> >>> I hope someone can help me with a problem I'm having with Ci

Re: [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image

2009-10-03 Thread James Stocks
On 3 Oct 2009, at 16:37, Jonathan Thurman wrote: > On Sat, Oct 3, 2009 at 6:17 AM, James Stocks > wrote: >> Hi everyone, >> >> I hope someone can help me with a problem I'm having with Cisco 7940 >> phones on the SIP 8.12 image. When I place a call fr

[asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image

2009-10-03 Thread James Stocks
Hi everyone, I hope someone can help me with a problem I'm having with Cisco 7940 phones on the SIP 8.12 image. When I place a call from one of the handsets, the call proceeds as normal for 20 seconds and is then terminated by Asterisk (1.4.26.2): [Oct 3 10:08:55] WARNING[1650]: chan_sip