similar to: Re: Asterisk-Users Digest, Vol 8, Issue 229

Hi, all ! I have a situation like this: [SIP Terminals] <-> [*] < -ISDN-PRI-> [Panasonic D500] <-> Telecom (conn to Telecom is with second PRI card in Panasonic and 16 POTS lines). Panasonic has 2 ISDN PRI cards (one to Telco, and second to Asterisk), 16 POTS lines to telco and 32 (advanced hybrid telephone type) extensions. Idea is to have possibility to have some users on

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

Thanks for the information Lee ! Still, something is still strange to me, since this two Panasonic Fax machines are receiving at least 20 multi-paged faxes a day (they are in same office as both Asterisk boxes, and me :) ). Beside that, POTS lines in those two faxes are from same PSTN switch as line in X100P in one of Asterisk boxes, and the ISDN PRI lines in other Asterisk box is from same PSTN

Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that

Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk "unallocated number" but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16

Hi all ! I'm trying to make a automonitor generated filename to "make its way" into CRD(usrefiled), so I can keep track of recorded conversations in CDR logs. Looking how to do that, I have found cool (but almost undocumented) option of res_monitor: if you set monitor format in form of "format:<string>" (i.e. "wav:monitor"), res_monitor will prefix the

Hi ! On Cisco 7960 (with or without 7914 add-on module) when I press speakerphone button (or select line with line button - which automatically put second line on speakerphone) after about 15-20 seconds of dialtone Asterisk stable dies (seg fault). Tested versions of Asterisk are 1.0.2, 1.0.3 or 1.0.5, chan_sccp is newest form CVS of chann-sccp.sourceforge.net ). Firmware of 7960 is

Hello ! Is there something wrong with the function HINT(<extension number>,n) ? Note the param n - it is supposed to get (as far as I understand the documentation) CALLERID(name) of the extension (via extension's hint). Example configuration: pjsip.conf: [10] type=endopoint . . callerid = Test extension <10> . . extensions.conf: [exts] exten => 10,hint,PJSIP/10 exten =>

Hi ! Does anyone managed to send multipage faxes (in single TIFF file) with app_txfax from spandsp package (i'm using 0.0.2pre18, libtiff 3.7.1)? If so, I'm interested in format of TIFF file that has been sent sent succesfully (tiffinfo <fax-filename>). I'm having problems with app_txfax, sending first page successfuly 99 % of the time, but never managed to send second or

Yes and No The D500 is a terrible thing First problem: it sends some horrible DTMF, so if your voicemail is configured to send #H and *H, it will not work, configure it to send numbers, like 8H and 9H (H is a placeholder for the extension). I also managed to use the MWI (message light), it's a perl script that is in voip-info.org, but with a little correction because the wiki distorted it. If

> -----Original Message----- > From: Eric Wieling [mailto:eric@fnords.org] > Sent: Friday, March 11, 2005 1:52 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: Incoming echo cancel > > > Nenad Radosavljevic wrote: > {clip} > > > > Anyone have an idea, why this type of echo happens ? As far as I have

Anyone have any luck with connecting Asterisk to the Panasonic KX-TD500. I have Asterisk connected via crossover to the TE110P. We are able to make internal calls into the Asterisk Box but the PBX vendor (I know nothing about the KX-TD500) tells us it is not possible route DID over the trunk. I find this hard to believe. Anyone have any luck with this? Thanks! -------------- next part

2014-02-06 11:09 GMT+01:00 giovanni.v <iax at keybits.org>: > Il 05/02/2014 8.42, Olivier ha scritto: > > channel then it depends upon what you have the priindication option >> set to. With >> priindication=outofband then a busy cause code is sent to the >> network and the call >> is hung up. With priindication=inband then a busy tone

On my current instance, the meta_data.json is the following: { "availability_zone": "nova", "devices": [], "hostname": "ims-host-1", "keys": [ { "data": "ssh-rsa

Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sending the call to Congestion() if no of calls in this group are more then 3. But my

Guys (and Gals), FYI I also have the *same* message here. Wonder is it is related to my Compaq D500 Space Saver PIV 1.7 or the fact that I don't yet have a modem card in the * box. (Please don't shoot me, did try Google first) Many thanks, Wagner Gimenes -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf

This one has compiled cleanly ! Thank you very much. Nenad > Oops, I sent the wrong one. Here's one I modified to work with 1.0.X > Try again > >> Nope ! This is the one that tries to include PRE 1.0.X header file >> <parking.h>. >> >> It cannot compile on * 1.0.X (I have tried also to include >> <features.h> instead of <parking.h>

A theoretical way to test this against any other protocol is to have a firewall rule which can be triggered locally which just drops the traffic and then re-enables the traffic in a while (short or long, can be decided/tweaked). This way, there is no need to think about which protocol is used, when we kill the connectivity. The only consideration could probably be that the dropped connections

I have an update on the networking issue: - After the deep dive into the logs of the firewall by customer's security team, it turns out that even though there were some disconnections, the time-stamps do not match. This means that we got the disconnected by something else (ESXi or conversion host perhaps) - As we mentioned in the chat briefly, there could be general keep-alive issues on both

Hi, I have asterisk connected to PSTN via H.323 gateway via chan_oh323. Incoming calls to SIP extensions work, but SIP message "486 busy here" from a busy extension isn't correctly forwarded to H.323. As a result, a caller from the H.323 side calling a busy SIP extension gets some rings and then an irritating timeout with H.323 message 'no user responding' instead of