search for: izmedia

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 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

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, 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

...ying 'vm-toforward' (language 'en') When trying to check vm... Anyone else seen this behaviour? We havent' made any changes to the server in months & I'm only now getting these reports. Thanks as always! Dan -----Original Message----- From: Dan Elder [mailto:isbeen@izmedia.com] Sent: Thursday, May 11, 2006 1:36 PM To: 'asterisk-users@lists.digium.com' Subject: Re: Voicemail problem, not playing back 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 &amp...

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

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.

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 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

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... 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

Hey all, guess I didn't include enough info in my last query. I'm having massive echo problems on our sip<=>fxo connections & have been reading up on echo cancellation & such. When I try to run fxotune I get the following error: Could not fill input buffer Tuning module 25.Failure! The system has two of the digium 4port fxo cards & one T1 card (which isn't fully

Is it possible to adjust the rx/txgain values per device? I have a mishmash of different phones (soft phones & sip hard phones) and each of them sound different w/the same rx/txgain settings. Is there any way to adjust these via asterisk? I'm having the most difficulty with sjphone & these zultys zip2 phones, vol is right on one, but way off on the other. Thx again...

Hello all, I'm working on asterisk fax>pdf/email & have a problem. I can see that the faxes are received (via the cli) and I get the fax pdf in my email, but they are always blank, any idea what is causing this? I'm using AAH 2.0 & have installed the fax/pdf (via install-pdf from the command line). Any ideas? Thanks Dan

hey all, a followup from yesterday, not only are my incoming faxes blank, but they are also EXTREMELY small (like .4in wide), I've seen several people mention this problem in my searches, but no definative answer that works with the fax>email setup. Is there any resource that explains how AAH handles this or some tips for troubleshooting this issue? I've gotten several replies to my

I've also purchased their GUI and hoped it would work for us, but the lack of proper documentation, horribly garbled tech support lines (support seems to come from Australia, and they apparently use very low quality voip trunks),broken installer, and cryptic interface forced me to reconsider. After hours and hours of wasted time, I chucked this product in the garbage in favor of AMP... wasted

Ok, got my last issue sorted, now another one. I can call out fine on this zap channel which is connected to a carrier access bank 1 channel bank, using asterisk 1.2 (aah2.0), I can call out, call other extensions & such.. but I cannot call into this zap extension, it always says the user is on the phone & asterisk -r shows "s-CHANUNAVAIL|1". In AMP does something other than the

Hi all, something odd is goin on w/my AAh2.0 install.. in my 'dial commands' section, I have Tt - but if I try to transfer a call I originated, the # key (attended transfer) nothing happens. I can transfer the call if its coming in, but not if I made the call... the dial commands seem to be set, but the feature isn't working.. is there an obvious place to look to fix this one? I'm