similar to: zaptel and low ring voltage

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

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, 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 -- ..................................

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

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 all, I'm experiencing a problem with SIP channels going ZOmBIE after the following sequence of events: - IAX2 client calls SIP client - SIP client consultive transfers (using sip REFER) the call to a MeetMe extension, and hangs up. At this point, the IAX2 client will indeed be in the meetme room, but a 'show channels' at the * CLI reveals that the SIP channels that were involved

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

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

I have a bind server running that cannot resolve www.atbfinancialonline.com. I turned on "debug 10" in the named.conf and start up dig on it, but dig just times out, what else can I do to see why exactly it won't resolve this? Thanks! jlc

Hi all, We have an analog line from telco, on which 3-way calling is subscribed to. This line is plugged into an FXO module on a tdm400p. If an incoming call comes in on this line, can */zaptel send Flash to telco via the FXO module? If it could, then an incoming call could be 'transfered' to a cell-phone, for example, with a single analog line. (where 'transfer' is really

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

Hi, We ran into a little problem recently with our phone provider (Telus Canada): we are unable to dial numbers outside North America. This is what happens: the phone number 011... is sent out over our T1, Telus sees the correct number on their switch. However the switch thinks it's a North American phone number (and thus has to have 10 digits) and rejects the overseas number as

i've managed to catch the exploding connections database in the act. i was able to attach gdb to the smbd while it was going crazy: (gdb) bt #0 0xf55da in tdb_expand () #1 0xf5787 in tdb_allocate () #2 0xf6554 in tdb_store () #3 0x6534 in claim_connection () #4 0x406a2 in make_connection () #5 0x1e105 in reply_tcon_and_X () #6 0x3e256 in switch_message () #7 0x3e8c7 in chain_reply ()

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

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

Hello All, As you can see below I am having a problem checking for updates. This happens repeatedly. I have to kill the process then rerun. I have tried "yum clean all" but no joy - the process hangs again on "extras" - see second listing below. Suggestions? Dave [root at cserver ~]# yum check-update Loaded plugins: fastestmirror, priorities Loading mirror speeds from

We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. The technician was stumped. I suspect there's something specific that I need to do to make it work, since many PBXs can do this. The switch is a

Ryan, from the console what does "zap show channel 1" or 2/3/4 in your case say. I have X100P's and I seem to be having similar sounding problems. I noticed that the above command shows the channel to be off-hook at all times when a phone line is plugged in. I don't know why or if it is a bug in the application reporting the status. dbc. Ryan Courtnage wrote: > On July 8,

Hi; I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940 Paging adaptor. This port on the TDM400P was connected to a 2500 Set and was working I just re-connected it to the Valcom (which is known to work on a Telco POTS line) and its not picking up. The Valcom docs say it need a minimum of 75 Volts at 20-30 Hz to recognize a call... So the question is what ring voltage

Hi - Does anybody know if an FXS generates line voltage when Dahdi/Zaptel is disabled? Thanks, Noah