similar to: max queue time; newbie question (fwd)

I can't seem to make the fax detection work. Here's an excerpt from zapata.conf: signalling=fxs_ks group=0 context => guestaccess channel => 47-48 and from extensions.conf: [guestaccess] include => incomingmain [incomingmain] exten => s,1,Dial,Zap/1&Zap/9&Zap/10&Zap/11|24 exten => s,2,Voicemail,u7000 exten =>

<MOD NOTE:Please kill/bounce my other email, it was accidental.> I just pulled down the newest CVS and recompiled. FWD (free world dialup) works now from *, and I AM behind a NAT. I've nearly given up on the xten lite, iaxcomm sounds better. I'll be trying the other win app thats up-and-coming on the list later. It seems to have broken iptel, but that's not as important to

I don't know if 911 uses caller ID or BTN (Billing Telephone Number) 900 calls, operator calls, and 800 calls use the BTN not the Caller ID... Anyone???? 3. Re: E911 and asterisk (Martin Pycko) Message: 3 Date: Mon, 21 Jul 2003 12:05:38 -0500 (CDT) From: Martin Pycko <martinp@digium.com> To: <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] E911 and asterisk

Thanks for all your help Martin, Guys, This is a good find and hopefully could help someone else. I've been having a problem with forwarding voicemail from one mailbox to another. I ran down the sendmail and soundcard path and came up goose eggs. With intuitive guidance from Martin Pycko (Digium), I switched from Redhat 9 Kernel linux-2.4.20-8 to Redhat 8 Kernel linux-2.4.18-14 and it

Another issue to consider is T1 framing. If your application is putting bits onto the T1 at the rate of 1.544 Mbit/s then the T1 would need to be unframed. I don't believe this is an option in zaptel! If however, it is putting bits on at a rate of 1.536 Mbit/s and adding 8000 bit/s for framing then you may be able use the suggestion below. Don Pobanz On Thursday, April 03, 2003 3:28 PM,

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

Hi list! I'm having some problems with chan_sccp and a Kirk IP600. Basically the handsets work (they emulate a Cisco 7940) but I have the following issues: 1. If a handset is in a conversation and there is a new incoming call, the incoming audio is muted (but the other party can still hear anything spoken on the handset). What is normal Asterisk behaviour, that a handset is left alone

Hello, I created a new application for myself called Dialtone() by modifing res/res_indications.c file. It can be used as such: exten => s,4,Dialtone(30|${CALLERIDNUM}) exten => s,5,Playback(time-exceeded) exten => s,6,Goto(s|1) It will stutter if you have new voicemail and you have passed the mailbox number as I did above. It will stop dialtone the moment you press a key

Hi, After playing around with queues a bunch on 1.4.26.2, I noticed a few things, which the patch below addresses. It addresses: - Callers in position 0 will hear periodic/position announcements at a very different rate than all other callers. -- Announcements while in position 0 could be delayed up to "timeout+retry" seconds. -- This patch reduces that possible delay to only

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

I've got an * system that is having some real problems with 1.0.2. The biggest problem is that calls going through my T100P get choppy for about 10 seconds every 1 or 2 minutes. Asterisk is running on a debian stable system with current packages. The T100P is plugged into a Adit Channelbank with 8 POTS lines hooked up to the Channelbank. I've watched the vritual memory and CPU status on

Hi folks, My stack of trusty FreeBSD servers always seems to be growing, and it's getting to the point where the daily and security output mail is too much to make good use of. I'm looking for suggestions for log monitoring and aggregation tools, especially from a monitoring-for-security perspective. If I had to imagine an ideal system, it would be a central server that securely

HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070603/6564c117/attachment.htm

Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11

[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device state of this queue member, SIP/3617001000, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. [Cary Fitch] We are running 1.4.22 and this message popped up in console. It could be causing our Queues announcement problem, because if all members

The Asterisk Development Team has announced the release of Asterisk 11.22.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.22.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Is it possible to specify a 'wrap-up' time in a queue so agents will have a specified amount of time to complete tasks between calls unless they hit a key on the phone? As it is they can recieve a call moments after they hang up with no 'down time'. Thanks Jim Friedeck

Is there any way to receive and send faxes using a T100 card? If so how is it done? Gene Kochanowsky Solution Sciences, Inc.

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

I've read about securelevel in the mailing list archive, and found some pitfalls (and seems to me to be discarded soon). But According to me, the following configuration should offer a good security: - mount root fs read only at boot; - set securelevel to 3; - do not permit to unmount/remount roots fs read-write (now it is possible by means of "mount -uw /"); - the only way to make