similar to: send Flash via FXO

Hi all, I curious to know if other UIP200 users have this same issue: You flash (XFER button) to consult-transfer a caller to another extension. If the transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- ..................................

Hi all, A while back, there was a short thread on using the FXS interface from a Primus TalkBroadBand device (a DLink ATA) as a incoming line for the FXO interface on the TDM400P: Primus <--> DLink ATA FXS <--> TDM400P FXO <--> Asterisk In that thread, a couple of people suggested that this does not work reliabley, and the ATA FXS <--> TDM FXO link 'goes

I couldn't be happier with the simplicity of this - but it's a hack! Hi all, I'm currently using a SIP client (BT101) to connect via DSL to a remote instance of Asterisk. - Asterisk has a private IP behind my OFFICE router. - The SIP client has a private IP behind my HOME router. I'm doing this _without_ the use of STUN or proxy servers. Here's how it works: -

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). ?We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. ?Attempts to send outbound calls on any Zap channel will result in hearing a loud

Hello, We are using a TDM400p with 4 FXOs and SIP phones in a high call-volume environment. At least twice a day there are complaints of 'dropped calls'. Examining the debug logs, I see that in each case, an "on hook" event is detected, followed by the zap channel being hung-up and * saying "BYE" to the sip phone: Jun 23 14:17:22 DEBUG[2441232]: Exception on

Hello all, The "Secret Agent" final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script

Hi all, I've run into 2 separate problems relating to fax: 1) Using tdm400p + fxo, Asterisk is unable to detect the fax from some fax machines (from others it can). Using zap barge, I can confirm that these troublesome calling fax machines _do_ send the fax tone loud and clear. Are there certain circumstances where I should expect a Digium card to fail in detecting a fax? 2) Using

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

Hi all, Several months ago we built an * box with a quad-FXO tdm400p (REV e/f). >From the get-go, there has been a problem where occasionally (2-3 times a week) zaptel/* will not detect the ringing on a line. (The call will ring through to telco voicemail). The problem is not specific to a single line or FXO port on the tdm400p. I have 2 theories: #1 - the ring voltage for some calls is

Hello, I'm planning to convert my phone system to Asterisk, as I've outgrown my TalkSwitch system. I have a few questions for experienced * users, most of which can be answered yes/no. Current Setup: - Talkswitch 48NLS (4CO/8Ext) phone system. - One CO line, two Vonage lines, one Voicepulse line connected to phone system - A third Vonage line directly connected to a fax machine - A

I was wondering if anyone is able to use the three way calling features from their telco on the incoming FXO lines to transfer a caller back out to say a cell phone. I am currently moving from a Talkswitch to the Asterisk PBX and one nice feature they have is after 4 rings or so I can have the call transferred to my cell phone using the same line it came in on with three way calling. Just

Been using a few different FXO interfaces (X100P, Voicetronix, and SPA-3000) at a couple sites and they've all run into nasty issues at times. I'm looking at moving one of my offices and am in the market for a solution for connecting 4 FXOs. I got excited about recent offerings from Digium and Sangoma in the way of new cards but I hear they're causing troubles for more than a few.

I've been using my Asterisk (At my house - 2 modem-type fxos, and an assortment of SIP endpoints for phones) for about 5 weeks now, and I've been really happy with it, but I'm still having an echo problem that I've exhausted google with, and can't get straight... I think I've determined that because I'm using $7 voice modem clones for my FXOs that bad echo is going to

Hello all, A new version of the Asterisk Management Portal is available for download. Please visit the AMP homepage at http://amp.coalescentsystems.ca Upgrade instructions are at http://amp.coalescentsystems.ca/UPGRADE Use our Sourceforge mailing list and forum for discussions about AMP. 1.10.006 ChangeLog: - Use extensions_custom.conf for customizations. Sample included. - Added option

Hello all, The Asterisk Management Portal (AMP) is now known as FreePBX. FreePBX 2.0.1 is now available for download. A **BIG** thank you goes out to the project developers for all their hard work, and to beta testers for running FreePBX through it's paces! This exciting new release boasts a better user experience, additional functionality, and a new module system. The module system is

I compiled libtiff version 3.6.1 and spandsp and spandsp version k. When trying to load asterisk I get the folloein error: Jun 7 10:15:03 WARNING[16384]: loader.c:408 load_modules: Loading module app_dtmftotext.so failed! Ouch ... error while writing audio data: : Broken pipe [root@zapata root]# Warning, flexible rate not heavily tested! Please help! -- Manuel Marin Garcia TRANSTELCO S.A.

I have Asterisk CVS-HEAD-05/25/04-17:13:22, Copyright (C) 1999-2004 Digium. Usign exclusively digium hardware. 3 TDM400P cards. 1 4xFXO 1 4xFXS 1 1xFX0 & 3xFXS When * is attending FXO calls, bridged to FXS calls, natively ofcourse, at a random time, the call hangus up. Also, for example, if a call is done, and an other extension hangup, there are some probability that the other extension

Does anyone know how (if possible) to do three way calling on the UIP-200? There doesn't seem to be much info about this phone, but all the feature lists I've read says it can do conference calls. I can't seem to do it, though. Any help would be appreciated. -- Seth "et lux in tenebris lucet" Mattinen sethm@rollernet.us

I have a TDM400P card all installed, and I can call out from a sip phone, but when I call from the PSTN into the asterisk machine, as soon as the Answer() gets called, the dial tone changes and is sounds like there is a lot of static on the line. Below is the part of the dial plan for answering the call. exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,Dial(Sip/pfriedel,20,tT)