similar to: Modifying Voicemail menus?

I have a ring group set up with 3 extensions. we'll use 14, 15 and 16. When a call comes in, it rings all three extensions. If one particular extension already is on the phone, it completely skips that phone and only rings the other 2. Example to explanation sake is: Call comes in, ext. 14 is already in the middle of a call, 15 and 16 will ring normally, but 14 does not have any

I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD

We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Airlines (8004337300) and Staples (800-378-2753) - we can call many other 800 numbers just fine. Our asterisk setup has a 4-port

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most

Following up with some more tests... I just left a message in a voicemail box, and got the same behaviour (i.e. it says its starting to play ('First Message') then it goes to the VM ending menu (i.e. press 7 to delete) without playing the message. On this particular test.. I went ahead & left a 2nd VM in the inbox, then when I went to check them, they would both play properly... The

Hi again, I'm trying to setup our non pri T1 (they call it a Long Distance T1), our current pbx has the signaling set to E&M, I can set em in zapata.conf, but I'm trying to track down the proper entries for the zaptel.conf file. The digium docs only show a PRI example. Our current system has these settings: Signalling: E&M Framing mode: ESF Line Coding: B8SZ here's my

Anyone gotten two of the 2572 echo canceller cards to work in a 253c mounting assembly? I can get one to work, but when I install two, one always fails. I've tried all my cards solo in the enclosure, on each side, and they all work properly when only 1 is installed, however, when I install two, one of them will come up, but the other always fails. Anyone know what might be causing this?

_____ From: Cullin J. Wible [mailto:cullin@emailanalyst.com] Sent: Monday, July 25, 2005 3:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; 'asterisk-dev@lists.digium.com' Cc: 'David Rule' Subject: Voicemail Send Message (Options 3, 5) Patch We run Asterisk 1.0.9 with multiple voicemail contexts and realized that there was a bug in the Voicemail

30 says it's view only in the docs & I can't seem to change it, any other options? > Option 30 allows to set Module Shelf Address/ID.

Hey all, been noticing some oddness on a new AAH install... occasionally an incoming zap line with automatically connect with an outgoing extension, even though the incoming line hasn't specified what extension it's aiming for (i.e. haven't tapped in the ext # yet)... so someone's trying to call out from inside the office & are automatically connected with an incoming line.

Hey all again, I'm wrestling with echo problems on our sip extensions. I've set these items in zapata.conf but tweaking these values doesn't seem to make much difference echocancel=yes echocancelwhenbridged=yes echotraining=2500 rxgain=8.0 txgain=1.0 are there other settings that can help me tame this beast? Been searching but not turning up anything that'll work here. Thanks

Hello, >From dialplan perspective, it seems you can't tailor your voicemail behaviour to specific needs ("dial 1 for old message listening", ...). Can anyone recommend a way to do it ? Does it make sense to write your own IVR and store audio files somewhere ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, been searching & not finding an answer to this, although I'm guessing it's absurdly simple... I just hooked up a T1 to our * box (1.2.0), which had been using POTS lines via a channel bank.. Now when I call the new T1 circuit, there are no rings, the Autoattendant just picks up right away.. Any clue on how to make it ring twice before getting picked up? I tried immedate=no and

hey all, trying to get a zap extension to work & I can dial out normally with it, but if I try to access any of the features (i.e. *97 for voicemail) the zap channel doesn't hear it, and all i get is dialtone. Is there a dialplan setting or something to make the zap channels recognize keys like * or # ? Thanks in advance

Is there an easy way to disable flash transfers? I'd prefer the users hit # to transfer, since some users are hanging up a call, then dialing another one without giving the handset enough time to actually hangup the call, so it appears that they are transfering the 'ended' call to the new number that they are calling.. I'd like to keep flash functionality for call waiting, but

Hey all, is it safe to run the asterisk-update.sh script that comes with AAH to upgrade only the asterisk binaries? Doug has chimed in a few times saying 'upgrade' when I post problems, but Aah makes this really painful. I'm using AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in my installation. Can I safely upgrade just asterisk and not any of

Hi all, using * 1.2.9.1 and this week all of the sudden calls keep ringing even after they've been picked up... Here's one users summary: When I pick up the phone, I hear a dial tone and I am able to dial out. But for some odd reason, the receiving line picks up while the outgoing line is still ringing. And the receiving line can hear everything while the phone is still ringing. I tested

I'm running a recent CVS build under Solaris 10. In the shell than I'm running the Asterisk console I have TZ=US/Eastern and in my voicemail.conf I have tz=eastern and eastern=America/New_York|'vm-received' Q 'digits/at' IMp. The voicemail envelope information seems to be exactly 4 hours ahead. No matter what I try I can't seem to find the cause. Any ideas? Frank

Hi, all I am trying to set up voicemail. I've done it to the point where I can leave messages. How do I retrieve them? Actually I have few questions: 1. I want voice mail to be available at certain extension, say 100. How do I set it up so all users can call this number and get to their respective mailboxes. 2. How do I let users to create their own voicemail passwords from the phone? 3.