similar to: Calls keep ringing after being picked up

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

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

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

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

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

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

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? Doug.

* Is there a way to configure NUT so that it recognizes the huge additional battery pack that has been attached to the 2200XL? The 51minutes runtime is a number consistent with the load we have placed on a stock standalone 2200XL. With the extra large battery pack we've added, the runtime should be more than 2.5 hours. * I see the "Bizzare configuration" graphic on the

In the rdoc there''s an example: xml.em { xml.b("emp & bold") } Which produces: <em><b>emph &amp; bold</b></em> But what if I wanted to nest tags with text, i.e.: <em>just emph and <b>emph &amp; bold</b></em> ? I''ve messed around with it and have only gotten bizzare results or nothing displayed at all -

Hello all! I guess it's a standard reason for joining a mailing list, but I have a problem. For some reason, whenever I try to start dovecot, be it from the xinet.d or from the commandline, I get this rather bizzare error. I can't even work out what it means, so I can't even begin to debug it. There is nothing extra in /var/log/messages, so I'm out of ideas. I've tried it

Check this out. I recently closed a bug I had written, #495 "ExtraChannel in transfer causes crash" Now I've been able to reproduce it, and somewhat narrowed down the culprit. But before I write another bug report, I wanted to see if anyone else had experienced the following (or would like to try:) When DNS (or outside connection to the network, not sure which) is broken and

All, Using SIP with progressinband=yes I get Asterisk to generate the ringing sound for callers. However, I was wondering if it is possible to configure what is 'played back' to the calling party? i.e. instead of just 'ring ring' could I potentially play back a song from an MP3, WAV or GSM file? I'm thinking it would be quite cool to offer a customised 'ring'

Hello, My target system is : PSTN <---> Sip Provider <---IP/ADSL---> Router with fw/NAT <--- SIP/IP/Eth --> Asterisk <--- SIP/IP/Eth --> SIP Phones Asterisk is configured to keep NAT connection with SIP provider open (with qualifyfreq) so I don't have any problem (yet) with either casual incoming or outgoing calls. To work around a possible No Audio when an incoming

I just migrated from UW-imap to dovecot last night. After some tweaking of the dovecot.conf file, disabling xinetd's entries, firing up the dovecot daemon, and copying the .mailboxlist to .subscriptions for all users, things looked to be going just fine! I received a call this morning from a user stating that they had all of *my* emails in *their* inbox! They don't know when it

I'm using Rcmdr version 1.4-10 with R version 2.9.0 under Vista x64. A standard editing convention under Windows is that alt+backspace is a keyboard shortcut for "undo". I often find myself hitting the [alt] and [backspace] keys while editing scripts in Rcmdr. However, this causes the Rcmdr GUI to go into a crazy mode where the menu keeps flashing and the GUI is unresponsive.