Re: [asterisk-users] Dahdi/callerid issue

2010-01-19 Thread Remco Barendse
On Mon, 18 Jan 2010, ev...@disruptor.nl wrote: Hey Ira, It seems after a several testing, that the wait(1) seems to solve the issue. Only now weirdly enough the phone keeps ringing if the caller hangs up before i picked up the phone (pstn call) Regards, Evert Hi Evert I have been

Re: [asterisk-users] Dahdi/callerid issue

2010-01-19 Thread evert
Hey Remco, exten = s,1,Wait(1) exten = s,2,Answer() exten = s,3,Dial(SIP/phone) A bit more simplified then what i have in my config, but exactly the same order. Regards, Evert On Tue, 19 Jan 2010 10:46:52 +0100 (CET), Remco Barendse aster...@barendse.to wrote: On Mon, 18 Jan 2010,

Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]

2010-01-19 Thread listu...@spamomania.co.uk
On Wed, 2010-01-13 at 11:13 -0500, Kristian Kielhofner wrote: On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: ..snip.. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our

Re: [asterisk-users] Sipgate DTMF not detected

2010-01-19 Thread joern
listu...@spamomania.co.uk wrote: I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to recognize digits pressed on a keypad coming in from a Sipgate trunk. There answer was to set this: dtmfmode=rfc2833 in the general section of sip.conf This has made no difference.

Re: [asterisk-users] Sipgate DTMF not detected

2010-01-19 Thread listu...@spamomania.co.uk
On Tue, 2010-01-19 at 13:15 +0100, joern wrote: listu...@spamomania.co.uk wrote: I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to recognize digits pressed on a keypad coming in from a Sipgate trunk. There answer was to set this: dtmfmode=rfc2833 in the

Re: [asterisk-users] caller getting cut off intermittently

2010-01-19 Thread John Taylor
Hi, I've now set dtmfmode=rfc2833 and that seems to have fixed it John 2010/1/7 John Taylor j...@vetsurgeon.org.uk: We're now getting this problem on outgoing calls. I've forced the port to 100FD but still no joy. Anyone any ideas how to debug this- have added verbose to logger.conf

Re: [asterisk-users] caller getting cut off intermittently

2010-01-19 Thread John Taylor
Hi all, I've now set dtmfmode=rfc2833 instead of inband and that seems to have fixed it John 2010/1/4 John Taylor j...@vetsurgeon.org.uk: I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our

[asterisk-users] test case with queues and system()

2010-01-19 Thread Евгений Шишкин
Hello, list. First of all i want to say sorry for my english. Long story short, on my future work i'll deal with asterisk and now i have a test case. But i'm very young to asterisk and don't have a lot of time so any help is appreciated. Test case: We have e1 trunk and multi-channel sip line.

Re: [asterisk-users] wav to gsm can't play

2010-01-19 Thread Danny Nicholas
Just a WAG - Playback is freaking out because you have the wav and gsm file there concurrently. Remove the wav file and try it again. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zhang Shukun Sent: Tuesday,

[asterisk-users] Detecting incoming faxes and forwarding to phone fax machine

2010-01-19 Thread Jeremy Winder
I'm having a problem receiving incoming faxes and I'm hoping someone here can help me out. My system is a PBX in a Flash with one dahdi card for my incoming analog lines and another dahdi card for my analog devices (fax and modem). My dahdi-channels.conf file looks like: ; Autogenerated by

Re: [asterisk-users] How to escape characters in Dialplan

2010-01-19 Thread Dominik
Hello, excuse my unknowing question, but how can I open the dialplan in hex-mode? What file is it? Normaly, my dialplan ist read in textmode from /etc/asterisk/extensions.conf. Sure, I can insert a new line with a hex-editor into the textfile, but then the line got wrapped: WARNING[1450]:

Re: [asterisk-users] MeetMe Conferencing - Announce your own join/leave to yourself and other conference members

2010-01-19 Thread Darren Sessions
Well, never mind on this (didn't get any responses anyways). I basically removed the meetme announcement options and wrote the functionality from scratch into my AGI framework along with an announcement queuing daemon that runs continuously every second in the background that generates a call

Re: [asterisk-users] How to escape characters in Dialplan

2010-01-19 Thread Danny Nicholas
This link might help http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dominik Sent: Tuesday, January 19, 2010 11:01 AM To: Asterisk Users Mailing List -

[asterisk-users] ast_queue_log to mysql asterisk 1.4 ?

2010-01-19 Thread William Stillwell (Lists)
I know in v1.6 its part of logger.c but I noticed this: https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=11625 However, it doesn't seem to ever been applied to any version of 1.4.x branch.. Nor can I figure out what it was applied to? This is over 3 years old, you

Re: [asterisk-users] ast_queue_log to mysql asterisk 1.4 ?

2010-01-19 Thread Dpto. de Sistemas
of figured it would have been applied to 1.4 at some point in time.. Any ideas? __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4786 (20100119) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com

Re: [asterisk-users] ast_queue_log to mysql asterisk 1.4 ?

2010-01-19 Thread William Stillwell (Lists)
? __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4786 (20100119) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com _ -- _ -- Bandwidth and Colocation

Re: [asterisk-users] ast_queue_log to mysql asterisk 1.4 ?

2010-01-19 Thread William Stillwell (Lists)
, versión de la base de firmas de virus 4786 (20100119) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com _ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Call drop-out on second incoming call.

2010-01-19 Thread Mike Diehl
Hi all, I've got 12 Polycom 430's behind a NAT that are working pretty well except for one thing: If one of my users is on the phone when another call comes in, there is about a 10 second time during which the user can't hear the call they're on. Then it returns to normal. My research

[asterisk-users] Initialize mailbox greeting message

2010-01-19 Thread Olivier
Hello, When a new mailbox is defined in voicemail.conf, it seems nothing is changed in /var/spool/asterisk/voicemail and the /var/spool/asterisk/voicemailcontext/vmid/ hierarchy is created whenever a message is dropped. Is this correct ? If positive, is it safe (ie is this a feature) to create

Re: [asterisk-users] How to escape characters in Dialplan

2010-01-19 Thread Olivier
I don't think my previous answer was very clear. Please, apologize for that. Some time ago, I also asked for the same feature as yours : to be able to send empty strings to IP phone Thomson ST2030. Tilghman Lesher from Digium was kind enough to code a patch for 1.4 (svn 187362). As

Re: [asterisk-users] How to escape characters in Dialplan

2010-01-19 Thread Olivier
2010/1/19 Danny Nicholas da...@debsinc.com This link might help http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText Yes but the trouble is you can't send empty strings with SendText. With this specific IP phone, you sometimes have to. Cheers -Original Message- From:

Re: [asterisk-users] How to escape characters in Dialplan

2010-01-19 Thread Tilghman Lesher
On Tuesday 19 January 2010 13:03:19 Olivier wrote: I don't think my previous answer was very clear. Please, apologize for that. Some time ago, I also asked for the same feature as yours : to be able to send empty strings to IP phone Thomson ST2030. Tilghman Lesher from Digium was kind enough

Re: [asterisk-users] How to escape characters in Dialplan

2010-01-19 Thread Dominik
Hm, (svn 187362) Permit zero-length text messages in SIP Maybe a zero-length text also works to clear the display of Thomson 2030: But with SendText it doesn't: -- Executing [...@intern:1] Answer(SIP/4711-0016, ) in new stack -- Executing [...@intern:2] SendText(SIP/4711-0016, )

[asterisk-users] How to enable a range of IP addresses in realtime sip_buddies

2010-01-19 Thread Bruce Ferrell
I'm using realtime sip peers and I need to enable a range of IP addresses for a peer. I have: deny = 0.0.0.0/0.0.0.0 permit= xxx.yyy.zzz.0/255.255.255.0 mask = 255.255.255.0 defaultIP = xxx.yyy.zzz.112 host = xxx.yyy.zzz.112 Addresses other than .112 are being denied. Can

Re: [asterisk-users] How to retrieve a phone number fromcall forwarding?

2010-01-19 Thread Soonthorn Ativanichayaphong
okay. Never mind. The redirecting number information is indeed available via ${CALLERID(rdnis)} variable. I was testing with two different phones and it happens that our ISDN provider is having problem sending RDNIS information on one of the phone that got me confused. Thanks. On Tue, Jan 19,

Re: [asterisk-users] How to enable a range of IP addresses in realtime sip_buddies

2010-01-19 Thread Jonathan Thurman
You need to set: host=dynamic Otherwise only .112 is allowed. -Jonathan On Tue, Jan 19, 2010 at 1:17 PM, Bruce Ferrell bferr...@baywinds.org wrote: I'm using realtime sip peers and I need to enable a range of IP addresses for a peer. I have: deny      = 0.0.0.0/0.0.0.0 permit    =

Re: [asterisk-users] How to enable a range of IP addresses in realtime sip_buddies

2010-01-19 Thread Bruce Ferrell
Thank you! Jonathan Thurman wrote: You need to set: host=dynamic Otherwise only .112 is allowed. -Jonathan On Tue, Jan 19, 2010 at 1:17 PM, Bruce Ferrell bferr...@baywinds.org wrote: I'm using realtime sip peers and I need to enable a range of IP addresses for a peer. I have: deny

Re: [asterisk-users] How to escape characters in Dialplan

2010-01-19 Thread Olivier
2010/1/19 Dominik d0m1...@geekmail.de Hm, (svn 187362) Permit zero-length text messages in SIP Maybe a zero-length text also works to clear the display of Thomson 2030: But with SendText it doesn't: -- Executing [...@intern:1] Answer(SIP/4711-0016, ) in new stack -- Executing