Re: [Asterisk-Users] Delayed ringing on some SIP phones

2005-10-17 Thread jurczak
Why dont you make a special extension where you could provide the delay and the numbers you want to dial? for example exten = _900X,1,Wait(${EXTEN:4:2}) exten = _900X,2,Dial(SIP/${EXTEN:5}) then in the incoming context you could dial exten =

RE: [Asterisk-Users] Delayed ringing on some SIP phones

2005-10-17 Thread jurczak
I am glad it helped, I hope the solution will not provide you any troubles with the CDR. Jurczak -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Monday, October 17, 2005 5:28 PM To: 'Asterisk Users Mailing List - Non-Commercial

RE: [Asterisk-Users] Will pay for asterisk help...

2005-07-12 Thread jurczak
It would be better if you could give us some more details about your configuration so that someone could help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 12, 2005 3:12 PM To: asterisk-users@lists.digium.com

Re: [Asterisk-Users] Sharing variables between contexts

2005-07-11 Thread jurczak
I am having the same thing on my extensions.conf and it works fine. I am using Asterisk 1.0.7 On Mon, 11 Jul 2005 12:04:59 +0200 (CEST), Armin Schindler wrote On Mon, 11 Jul 2005, Frank Schoep wrote: Hello all, I'm having trouble getting variables to work the way I want them to, let me

RE: [Asterisk-Users] Fw: linksys rt31p2 test case

2005-06-27 Thread jurczak
Are you pinging them from the same VLAN? Or a VLAN that has access to it? Marios -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dionisis Koumouras Sent: Monday, June 27, 2005 1:36 PM To: asterisk-users@lists.digium.com Subject:

RE: [Asterisk-Users] Accessing SIP username from AGI script

2005-06-27 Thread jurczak
Hello, Why dont you try using the the CALLERIDNUM or maybe the ACCOUNTCODE and based on that to take your decision? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Shirley Sent: Monday, June 27, 2005 4:52 PM To: Asterisk Users Mailing

RE: [Asterisk-Users] LiveVoip is Bankrupt

2005-06-27 Thread jurczak
I am throwing my * away, I will be using this system from now. Btw: I have laughed like this for long time -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Kalcevich Sent: Monday, June 27, 2005 10:28 PM To: andrew matthews; Asterisk Users Mailing

RE: [Asterisk-Users] cdr and billing

2005-06-26 Thread jurczak
Why dont you try nocdr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mahmoud Badran Sent: Sunday, June 26, 2005 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] cdr and billing Hello ; how can i

RE: [Asterisk-Users] CDR: source completed with sip domain

2005-06-26 Thread jurczak
From what I have seen, cdr does not add the domain, maybe you could use the userfield, otherwise you should change the source from asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rosario Pingaro Sent: Sunday, June 26, 2005 6:59 PM

RE: [Asterisk-Users] Asterisk 'losing' upstream provider registrationstate during small network outages.

2005-06-25 Thread jurczak
Some time ago (with previous releases of Asterisk) I had the same problem with broadvoice, so I added a cron job that reloads the sip every 1 hour. I know this is not the best solution, but at the time this seed fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]

RE: [Asterisk-Users] Voicemail

2005-06-24 Thread jurczak
What is your default context in the voicemail.conf Maybe you should try Voicemail([EMAIL PROTECTED]) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MDM Sent: Thursday, June 23, 2005 11:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [Asterisk-Users] Music on Hold Choppy

2005-06-23 Thread jurczak
do you have VAD enabled? On Thu, 23 Jun 2005 12:23:15 +0300, Mahmoud Badran wrote Hello all i am using asterisk 1.07 with mpg123-0.59r but still i get very choppy sounds, any suggestions? extensions.conf --- exten = 444,1,WaitMusicOnHold(120)

Re: [Asterisk-Users] Dialplan Q: Dialing with Capi

2005-06-22 Thread jurczak
You could try the new chan_capi-cm-0.5.1 in which you dont need to have any msn defined in capi.conf On Wed, 22 Jun 2005 12:28:58 +0200 (MEST), Patrik Schindler wrote Hello, I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI as channels. A call comes in via IAX2 and

RE: [Asterisk-Users] call file ignored?

2005-06-20 Thread jurczak
Hello, I just tried it, and it worked fine for me. Of course the context and the Extension where different. Is the Channel correct? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Monday, June 20, 2005 2:12 PM To: Asterisk Users List

RE: [Asterisk-Users] call file ignored?

2005-06-20 Thread jurczak
! On Mon, 20 Jun 2005, jurczak wrote: Hello, I just tried it, and it worked fine for me. Of course the context and the Extension where different. Is the Channel correct? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Monday, June

RE: [Asterisk-Users] ISDN Sub-Address

2005-06-12 Thread jurczak
Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ISDN Sub-Address On Sun, 12 Jun 2005, jurczak wrote: Hello, I have 2 Fritz!Pci cards installed on my system, and I use chan_capi. Recently I tried to work a little bit with sub addressing, and after searching

[Asterisk-Users] ISDN Sub-Address

2005-06-11 Thread jurczak
Hello, I have 2 Fritz!Pci cards installed on my system, and I use chan_capi. Recently I tried to work a little bit with sub addressing, and after searching the internet I did not find anything. Can anyone tell me if subaddress is in the called-party field just like the MSN? Or is it