similar to: VoiceMail fails to delete messages after emailing them

Anyway to make hitting `0` during a voice mail dial an extension? The bosses used to have that feature and love it. Their VM prompt would say: "Hello, My name is blah blah. I am currently unavailable. If you would like to speak to an operator press 0 now, otherwise leave me a message". Extension 0 exists, but dialing it during a VM prompt does nothing. Thanks, -- Daniel Jimenez

http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

Hi all. I've ordered a TDM400P with 4 FXO, but after using my X100P I'm thinking about returning the TDM400P because of bad echo issues. If I do get the echo issues I'll look at digital options. My question: Is anyone using ISDN (BRI) in the states? I've heard ISDN4LINUX devices suffer bad echo but chan_capi works great. All the chan_capi cards I find though are for overseas

I frequently get this error message, it repeats itself hundred/thousands of times and never stops. chan_sip.c:7467 sipsock_read: Failed to grab lock, trying again... During this period, I can make no SIP calls what-so-ever. The only way I've been able to stop it is to killall -9 asterisk. Doing a restart now doesn't respond. Anyone know why? -- Daniel Jimenez

Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has

I'm using Asterisk to registry several DDI's to a sip proxy (pipecall.com). Everything works fine apart from several times a day my firewall (zywall70) reports a UDP port scan attack from the pipecall sip proxy. I can't seem to work out why this should be. All I could think was that the sip registry was expiring and causing some strange probing from the proxy, is it possible to alter

I would like asterisk to dial an extension or external number but for the call to only be connected after the called party presses a key. Therefore been able to announce the call to the called party before answering. I have had this working on queued calls but want to incorporate this for standard dialled extensions. Our use for this would to be able to divert a call to a users mobile but only

Hello, I've recently purchased Adit 600 with 3FXS and 1FXO to be connected to my * server via T100P card. This is the output of "status equipment" command in the Adit600: For some reason the FXO card is seen as FXS, why? Is it ok? On the card it is written "FXO". Regards, Shlomi Bachar -------------- next part -------------- An HTML attachment was

Hello, I've been playing around with two generic X100P analog cards to create a proof-of-concept system before we go ahead and hook up a PRI. I'm running into a reproducible problem with sound quality of bridged calls, and am hoping someone will be able to point me in the right direction. I have in my dial plan a _9. extension so outgoing calls can be made... the first thing is

Hi, i want to use soxmix to record some calls in my PBX. If i use soxmix on my linux shell it works so i can mixed two calls into one consolidated call. I want to do the process automatically since extensions.conf but it doesnt work. My extensions.conf looks like this: exten => 407,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor) exten => 407,2,Monitor(wav,${TIMESTAMP}.${CALLERIDNUM}.wav)

Hi, We are having an issue here. It seems that whenever we initialize Asterisk on our network, the router that the Asterisk server is connected to (Cisco 7200) crashes and loses it configuration. This has happended five times and each time we have tested it, it is always when Asterisk starts up. Has anyone else seen this problem? It is very odd because this is a very good router and we

Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none

Dear All, I saw some topics discussed in forums about this, but none with my problems :( Currently we have a mailserver postfix with courier-imap-4.1.1, and courier-pop3 and courier-authlib. Our Maildir's are written in the users home directory (ex: ~/Maildir) I'm now migrating the mailserver to other cluster and installing with dovecot. We're also moving the Maildir's to

https://bugs.freedesktop.org/show_bug.cgi?id=92961 Bug ID: 92961 Summary: Xorg freezes (only mouse and ssh are still working) Product: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: critical Priority: high Component: Driver/nouveau

Dear all, I would like to have the same color for the all boxplots from the same panel, but my code below shows the two colors alternating. Thanks! set.seed(42) D1 <- rnorm(200) D2 <- factor(sample(letters[1:2],200,TRUE)) D3 <- factor(sample(letters[3:5],200,TRUE)) DF <- data.frame(x=D1,a=D2,b=D3) print(bwplot(b~x|a,data=DF,col=c("black","black"),

Buenas noches; Traigo una pregunta que supongo que alguno ya la tendrá resuelta, porque se me hace difícil entender algo que presupongo fácil. Quiero, según los valores de 3 o 4 variables numéricas, convenientemente escaladas, conseguir gamas de colores. Supongamos las variables numéricas: X, Y, Z; a cada variable le correspondería un color; pongamos que X = C (cian), Y = M (magenta) y Z = Y

Hi list! I think there are many german users in this ML, that use Asterisk with the new line of Deutsche Telekom (Magenta Zuhause). My ISDN will be converted in Juli (Kaboom-day at Juli, the 3rd...) and right now I can just hope, that I configured my Asterisk well to work with Deutsche Telekom, but I cannot be sure, since I can't test it... So my question: can someone using Asterisk with

Hi, all My Asterisk uses a Digium TE120Pand I would like to add an Astribank zaptel_hardware sees is, but I cannot get it working pbx:~# zaptel_hardware Argument "IRQ" isn't numeric in numeric comparison (<=>) at /usr/local/share/perl/5.8.8/Zaptel/Span.pm line 114. usb:005/002 xpp_usb- e4e4:1131 Astribank-8/16 USB-firmware pci:0000:04:00.0 wcte12xp+