similar to: Voicemail problem, not playing back audio

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

The wav file seems fine when I can catch it, I'm able to play it through winamp without a problem. Something is making * skip to the end of the message & not actually play any audio, then it moves the message to the 'old' folder, and agents don't normally check that folder. Any ideas where I can look to try to track this down? Thanks! Dan Elder wrote: > Hey all, am running

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, posted this previously, but didn't get a resolution... Maybe a rewording would help. A few times a week I will get a call from a user who has a new voicemail, but they cannot play it. They go through the menus, hit 1 to play the message, and immediately the 'post message' menu prompts them to delete the message. The actual voicemail file never gets played... This does NOT

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.

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?

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

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 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

I am new to asterisk and I'm setting up a test box to flesh out a switchover we're going to do at work. Right now I'm working on voicemail. I can leave a message fine, but when I attempt to listen to messages, I am having trouble. I can dial an extension for voice mail main, login, and it'll tell me how many messages I have. I press one and it will give me the date/time for the

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

Hi Guys Which module on Asterisk is the one in charge of playing the VoiceMail Server Audio to the end customer? I have work with MRFP but is it a module included in the SW? Need and external source? BR Joaquin This email is confidential and may be subject to privilege. If you are not the intended recipient, please do not copy or disclose its content but contact the sender immediately upon

Hi Madushan Maybe I was not clear ?. After SIP negotiation and SDP set up on the VoiceMail Server ?. Is there a file to specify a MGw (the machine that deliver RTP packages to end user)? From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Madushan Geethanga Sent: 15 July 2016 13:00 To: Asterisk Users Mailing List - Non-Commercial

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

Hello, I've just set up Asterisk with a TDM400P (specifically, a 'TDM22B bundle' - 2 FXS and 2 FXO, although I'm only using one of each interface at the moment, in case that's relevant) The problem I'm getting is that when I listen to a voicemail message, the recorded message is played back at extremely low volume. All the supporting prompts are at the correct volume,

Is there a way to edit the options available in the voicemail menu trees? My users are complaining that it's too complicated (I know, it's not really complicated), and I wanted to remove some of the options if this is possible. So far I havent' found any info on the wiki or searches, not that it isn't out there.. I just cant' seem to find it.. Any pointers? Thanks Dan

Hi All... Well, I'm still experiencing LOTS of dropped calls since installing the new (non pri) T1 here... I keep noticing a few things in the logs when this happens, namely the "Wink/Flash" statements and the "Didn't get a frame" messages... Anyone got any ideas on if this is a telco issue, a wiring issue, or an asterisk issue? Been trying to track this down via all 3