similar to: RE: Asterisk-Users Digest, Vol 23, Issue 182

Does anyone have an xml interface to comedian mail setup for the aastra 480i CT? The old interface to comedian mail on the ADSI phones was a nice feature although kind of slow loading. I thought I'd try to duplicate it on the SIP phones, but didn't want to reinvent the wheel. Marv Horst -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a "SAY NUMBER 123 #" before and after

Hi all, I am using SPC techniques of plotting Xbar, R, S, P and C charts in R.For this I am using qcc package in R and also using the following document for reference. http://www.stat.unipg.it/~luca/Rnews_2004-1-pag11-17.pdf But while plotting S chart, R chartchart and also C chart I am getting following errors. I have plotted the Xbar chart successfully using following command obj <-

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

Good day list, I hope, you will pay attention to my small problem (big problem for me). I'm totally new to samba, so I don't know exactly, where the problem is. Situation - I need to setup samba as a domain controller (and fast!). Server - openbsd 3.9, samba 3.0.21. There are my configs: smb.conf -------------------------------------------------------------------------------------------

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

_____ 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

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?

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.

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

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

Dear All I have jus migrated to centos 5 and jus wondering why n where this problem could be server A which is running for the past 2 years used as my primary dns and mail server redhat linux 8 sendmail ver-8.12.5-7 bind ver -9.2.1-9 since the server is old and is prone to all security problems i have a new machine which i have installed the following and will replace my existing redhat dns +

Dear All I have jus migrated to centos 5 and jus wondering why n where this problem could be server A which is running for the past 2 years used as my primary dns and mail server redhat linux 8 sendmail ver-8.12.5-7 bind ver -9.2.1-9 since the server is old and is prone to all security problems i have a new machine which i have installed the following and will replace my existing

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, 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 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