Re: [asterisk-users] Cisco 7940: showing FWD in display.

2010-02-15 Thread Michiel van Baak
On 21:40, Sun 14 Feb 10, Oliver Nittka wrote: Olivier schrieb: Thanks for the suggestion anyway, I'm going to test this just out of curiosity :-) And that's what i get in the CLI: Got SIP response 501 Not Implemented back from XXX.XXX.XXX.XXX Well, I guess I should really give

[asterisk-users] Zaptel/DAHDI error's on PRI

2010-02-15 Thread Sascha Ferley
Hi I've been running into a weird issue, which its hard to get any information on. We successfully setup a R710 system with Asterisk 1.4.22 / libpri 1.4.7, utilizing a Digium TE121B Pci express card. However we are having some stability issues and can't seem to trace it down to if it is a

Re: [asterisk-users] Call Pickup with 1.6.2.1 and Snom

2010-02-15 Thread RABOUIN Geoffroy
Hi, I've experienced the same thing in the 1.6.2 release, with the 1.6.1 all work as expected. There is nothing in the changelog ... So, I think it's a bug ? -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de

Re: [asterisk-users] Important security alert: update your dialplans now!

2010-02-15 Thread Lenz Emilitri
Or one could simply rewrite to: [incoming-from-voip] exten = XXX,1,Dial(${ext...@incoming-from-voip-old) exten = ,1,Dial(${ext...@incoming-from-voip-old) exten = X,1,Dial(${ext...@incoming-from-voip-old) exten = XX,1,Dial(${ext...@incoming-from-voip-old)

Re: [asterisk-users] Cisco 7940: showing FWD in display.

2010-02-15 Thread Oliver Nittka
Michiel van Baak schrieb: Or use the provided chan_skinny I tested chan_skinny some time ago, I remember it didn't work as expected, just can't remember what it was. Perhaps I should also test it again. Thanks! -- o -- _

Re: [asterisk-users] Important security alert: update your dialplans now!

2010-02-15 Thread Olle E. Johansson
15 feb 2010 kl. 09.33 skrev Lenz Emilitri: Or one could simply rewrite to: [incoming-from-voip] exten = XXX,1,Dial(${ext...@incoming-from-voip-old) exten = ,1,Dial(${ext...@incoming-from-voip-old) exten = X,1,Dial(${ext...@incoming-from-voip-old) exten =

Re: [asterisk-users] Important security alert: update your dialplans now!

2010-02-15 Thread Randy R
On Mon, Feb 15, 2010 at 9:51 AM, Olle E. Johansson o...@edvina.net wrote: To avoid extensive rewriting and fix the current issue. That works in countries where you have fixed-length numbers. Unfortunately, not every dialplan works that way, so that can't be a generic advice even though it

Re: [asterisk-users] Important security alert: update your dialplans now!

2010-02-15 Thread Rob Hillis
On 02/15/10 20:00, Randy R wrote: Olle, this may be a stupid question, but shouldn't a native santitize function be urgently added to the code base in all versions or change the dialplan compîler to ignore dangerous characters? Whilst I agree with this, the unfortunate attitude we seem to

Re: [asterisk-users] signal problem

2010-02-15 Thread Jeff Brower
Cool Dude- You keep asking the same question over and over, which is not cool. voice mail is working when ever call is received, extension 2000 receives it and if not answered in 20 secs, message is stored in voicemail no problem in that. after creating voice mail if some one again call

[asterisk-users] strange asterisk behaviour on XEN

2010-02-15 Thread Emre Kurnaz
Hi all, Now a days we are planning to run two asterisk boxes on XEN with DNS Failover. But even using the default configuration asterisk shuts itself down at least 5 times in a day with an exit status of 139 (i think it should be 139-128=11 there may be a coding mistake). Thus what do you

Re: [asterisk-users] Important security alert: update your dialplans now!

2010-02-15 Thread Olle E. Johansson
15 feb 2010 kl. 10.00 skrev Randy R: On Mon, Feb 15, 2010 at 9:51 AM, Olle E. Johansson o...@edvina.net wrote: To avoid extensive rewriting and fix the current issue. That works in countries where you have fixed-length numbers. Unfortunately, not every dialplan works that way, so that can't

Re: [asterisk-users] Still on spandsp/app_fax and T.38

2010-02-15 Thread Vinícius Fontes
- Kevin P. Fleming kpflem...@digium.com escreveu: Vinícius Fontes wrote: Will do. You guys will have my feedback on monday. If everything goes okay with that change, I'll post a patch on Mantis. No need for the patch; it's already on my radar, and if you confirm that it

Re: [asterisk-users] Important security alert: updat e your dialplans now!

2010-02-15 Thread Tilghman Lesher
On Monday 15 February 2010 03:37:24 Rob Hillis wrote: On 02/15/10 20:00, Randy R wrote: Olle, this may be a stupid question, but shouldn't a native santitize function be urgently added to the code base in all versions or change the dialplan compîler to ignore dangerous characters? Whilst

Re: [asterisk-users] Still on spandsp/app_fax and T.38

2010-02-15 Thread Steve Underwood
On 02/15/2010 08:57 PM, Vinícius Fontes wrote: - Kevin P. Flemingkpflem...@digium.com escreveu: Vinícius Fontes wrote: Will do. You guys will have my feedback on monday. If everything goes okay with that change, I'll post a patch on Mantis. No need for the

Re: [asterisk-users] Important security alert: update your?dialplans now!

2010-02-15 Thread Michiel van Baak
On 08:48, Mon 15 Feb 10, Tilghman Lesher wrote: On Monday 15 February 2010 03:37:24 Rob Hillis wrote: On 02/15/10 20:00, Randy R wrote: Olle, this may be a stupid question, but shouldn't a native santitize function be urgently added to the code base in all versions or change the

Re: [asterisk-users] SIP RTP ports not released when channel is hung up

2010-02-15 Thread Klaus Darilion
Am 13.02.2010 09:26, schrieb Olle E. Johansson: 12 feb 2010 kl. 16.43 skrev Klaus Darilion: Am 11.02.2010 21:09, schrieb Olle E. Johansson: 11 feb 2010 kl. 13.30 skrev Klaus Darilion: Am 11.02.2010 11:21, schrieb Armin Schindler: Hello, using Asterisk 1.4.28, I encountered a problem

Re: [asterisk-users] Important security alert: update your dialplans now!

2010-02-15 Thread Lenz Emilitri
Yes but in any case you can enter all of the strings that reasonably match - even if you have variable-length numbers, you will be able to determine that a valid number be between 5 and 15 characters - or likely 2 to 20, all numbers. A number of 156 characters is very likely to be a problem. BTW,

Re: [asterisk-users] Still on spandsp/app_fax and T.38

2010-02-15 Thread Kevin P. Fleming
Steve Underwood wrote: FFA sends its repeating no-signal and preamble packets with incrementing sequence numbers. While its not the only system which does that, it confuses some T.38 implementations. The T.38 spec is too vague to say whether the practice is right or wrong. In other

[asterisk-users] Maximum call handling capacity on single server

2010-02-15 Thread Amit Patkar | Avhan Technologies Pvt. Ltd.
Hi I have a server with Quad Core Xeon 2.4GHz and 4GB RAM. I want to use it for PSTN-IP gateway. What is the maximum call handling capacity I can achieve with this server? I want at least 480 concurrent PSTN-IP calls. That mean I will have to install minimum 4 x 4E1 cards and run 480 G.711 RTP

Re: [asterisk-users] Still on spandsp/app_fax and T.38

2010-02-15 Thread Steve Underwood
On 02/15/2010 11:27 PM, Kevin P. Fleming wrote: Steve Underwood wrote: FFA sends its repeating no-signal and preamble packets with incrementing sequence numbers. While its not the only system which does that, it confuses some T.38 implementations. The T.38 spec is too vague to say

Re: [asterisk-users] Still on spandsp/app_fax and T.38

2010-02-15 Thread Kevin P. Fleming
Steve Underwood wrote: In callweaver we gave the frames an extra parameter, which is the number of copies to send. It is the UDPTL code which creates the extra copies on the wire. They are not spaced in time, which would probably be a good enhancement. Packet loss tends to be bursty, so if

Re: [asterisk-users] Maximum call handling capacity on single server

2010-02-15 Thread Philipp von Klitzing
Hi! I have a server with Quad Core Xeon 2.4GHz and 4GB RAM. I want to use it for PSTN-IP gateway. What is the maximum call handling capacity I can achieve with this server? I want at least 480 concurrent PSTN-IP calls. It might be wiser to spread this over two servers or use external

Re: [asterisk-users] Maximum call handling capacity on single server

2010-02-15 Thread Steve Edwards
On Mon, 15 Feb 2010, Amit Patkar | Avhan Technologies Pvt. Ltd. wrote: I want at least 480 concurrent PSTN-IP calls. 0) Cross-posting is a no-no. 1) Not a -dev question. If you ever have any doubt a question belonging on -dev, it doesn't. 2) Putting 480 eggs in one basket is a recipe for a

Re: [asterisk-users] Important security alert: update your dialplans now!

2010-02-15 Thread Steve Murphy
On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri lenz.lo...@gmail.com wrote: Yes but in any case you can enter all of the strings that reasonably match - even if you have variable-length numbers, you will be able to determine that a valid number be between 5 and 15 characters - or likely 2 to

Re: [asterisk-users] Important security alert: update your dialplans now!

2010-02-15 Thread Tony Mountifield
In article 699ee941002150033t7c6e1be5xdba76cb0f68d5...@mail.gmail.com, Lenz Emilitri lenz.lo...@gmail.com wrote: -=-=-=-=-=- -=-=-=-=-=- Or one could simply rewrite to: [incoming-from-voip] exten = XXX,1,Dial(${ext...@incoming-from-voip-old) exten =

Re: [asterisk-users] Maximum call handling capacity on single server

2010-02-15 Thread Olle E. Johansson
15 feb 2010 kl. 17.36 skrev Steve Edwards: On Mon, 15 Feb 2010, Amit Patkar | Avhan Technologies Pvt. Ltd. wrote: I want at least 480 concurrent PSTN-IP calls. 0) Cross-posting is a no-no. 1) Not a -dev question. If you ever have any doubt a question belonging on -dev, it doesn't.

[asterisk-users] video voicemail

2010-02-15 Thread Jeff LaCoursiere
Playing around with the Grandstream GXV3140. I'm interested in having the video voicemail clips emailed in a format that might be opened by Windows Media Player or even Quicktime. Have been googling around a lot and have tried various bits of OSS to read the resulting .h264 file that

Re: [asterisk-users] video voicemail

2010-02-15 Thread Olle E. Johansson
15 feb 2010 kl. 20.31 skrev Jeff LaCoursiere: Playing around with the Grandstream GXV3140. I'm interested in having the video voicemail clips emailed in a format that might be opened by Windows Media Player or even Quicktime. Have been googling around a lot and have tried various bits

Re: [asterisk-users] video voicemail

2010-02-15 Thread Tilghman Lesher
On Monday 15 February 2010 14:09:38 Olle E. Johansson wrote: 15 feb 2010 kl. 20.31 skrev Jeff LaCoursiere: Playing around with the Grandstream GXV3140. I'm interested in having the video voicemail clips emailed in a format that might be opened by Windows Media Player or even Quicktime.

Re: [asterisk-users] video voicemail

2010-02-15 Thread Jeff LaCoursiere
On Mon, 15 Feb 2010, Tilghman Lesher wrote: On Monday 15 February 2010 14:09:38 Olle E. Johansson wrote: 15 feb 2010 kl. 20.31 skrev Jeff LaCoursiere: Playing around with the Grandstream GXV3140. I'm interested in having the video voicemail clips emailed in a format that might be opened by

Re: [asterisk-users] video voicemail

2010-02-15 Thread Tilghman Lesher
On Monday 15 February 2010 15:46:58 Jeff LaCoursiere wrote: On Mon, 15 Feb 2010, Tilghman Lesher wrote: On Monday 15 February 2010 14:09:38 Olle E. Johansson wrote: 15 feb 2010 kl. 20.31 skrev Jeff LaCoursiere: Playing around with the Grandstream GXV3140. I'm interested in having the

Re: [asterisk-users] Important security alert: update your?dialplans now!

2010-02-15 Thread Tilghman Lesher
On Monday 15 February 2010 09:05:33 Michiel van Baak wrote: On 08:48, Mon 15 Feb 10, Tilghman Lesher wrote: On Monday 15 February 2010 03:37:24 Rob Hillis wrote: On 02/15/10 20:00, Randy R wrote: Olle, this may be a stupid question, but shouldn't a native santitize function be

Re: [asterisk-users] Still on spandsp/app_fax and T.38

2010-02-15 Thread Vinícius Fontes
You could try defining the same identity string for app_fax that you have defined for FFA. Trying to make the other things more similar would require additional work. Maybe you should try that change first, as it is very simple, and requires no code changes. My receiving fax macro,

Re: [asterisk-users] Important security alert: update your?dialplans now!

2010-02-15 Thread Vinícius Fontes
He probably means AgentCallbackLogin While it has been deprecated, that hasn't been removed, either. If an enterprising person would like to try to fix it, I don't have an objection. Wasn't AgentCallBackLogin() removed in 1.6.1? --

Re: [asterisk-users] Important security alert: update your?dialplans now!

2010-02-15 Thread Tilghman Lesher
On Monday 15 February 2010 18:01:11 Vinícius Fontes wrote: He probably means AgentCallbackLogin While it has been deprecated, that hasn't been removed, either. If an enterprising person would like to try to fix it, I don't have an objection. Wasn't AgentCallBackLogin() removed in