similar to: DTMF recognized improperly?

We currently have a Merlin Legend system. The voicemail is falling apart (with the transition to a 10 digit timestamp on Sept. 8, 2001, the system locked up and refused to take calls; the official solution is to change the system time back to a year with a matching calendar). We are in the process of preparing the network infrastructure to support a VoIP system with Asterisk, but won't be

When I drop my file into the outgoing folder, the call is completed but the 'Context' entry is not respected. Instead, it drops into the default context. It does drop "properly" into the default context and function as would be expected. I looked through the source but didn't see any reason it would be completely ignoring the context. Call file: (where

swar sir, can u please unsubscribe me for your list b.regards jihad chalhoub --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, > visit > > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message

I have asterisk running as a voicemail system off of our Merlin Legend switch. We replaced our old Audix Voice Power (when the power supply fan died and burned it up) with asterisk a week ago. Many thanks to those who provided information about integrated VMI on the legend. The Audix system would, after a mailbox was closed, wait a few seconds, then use that line to dial the switch and update

Hi, First off, a big thanks to Digium (Mark, John, and Martin) for helping sort out a BellSouth config issue on our PRI. T100P working like a champ! Now it's back to tweaking the configuration on our SIP phones (7960s). The message_uri parameter in the phone's configuration file is working great. Dials comedian mail directly. Is there a way to let voicemail2 know what the incoming

Hi, anyone here is using HyperVM? As you probabily know, the owner of LxLabs has killed hitself (i want to make my condolences to his family): http://timesofindia.indiatimes.com/Bangalore/Techie-hangs-himself-in-HSR-Layout-/articleshow/4633101.cms HyperVM has some big vulnerabilities and we don''t know if they will be fixed. We don''t know if the licensing server will be kept

I'm in the process of setting up a test/demonstration system to show that VoIP is realistic and applicable for our needs. We put a 7905 and 7960 on a request for quote that went out the other day (to people like CDW & Microwarehouse). All of the vendors returned thier quotes without including the Cisco phones. So my question: where do you buy your phones? We can't buy direct from

Dear All. I want to become a wholesale VoIP traffic Provider , and i don't have a experience about the software used this career . I ask about Freeside billing system , FreeRADIUS AAA server and Asterisk telephony server gave me all i need to start my business . thanks -- Best Regards Mohamed Daif -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets "stuck" off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being

We seem to have stumbled onto an intermittant bug using smbclient to retrieve files from a W2000 server. Every so often it just ignores some files on a recursive mget. When it ignores files it always ignores the same files. The exact same command repeated again may (or may not) pick up that file. Running a DIR on the directory in question shows the same defect, sometimes the cursed file is

Hello. I've seen some mentions of asterisk possibly being used as an inexpensive voicemail attachment to a commercial PBX etc. Does anyone here, have experience of using it in this fashion ? What commercial systems have been successfully attached too ? How is the attachment made ? Analog, digital ? If anyone has successfully accomplished this, I would like to hear the make and model of

Hi I wonder if there's a smarter way to do this procedure. I have a vector of filenames wher I only am interested in the first part of the filename. I would use the following method of extracting the first part. But is there a more simple way of doing this? Names <-

I have two datasets: A with columns Open and Name (and many others, irrelevant to the merge) B with columns Time and Name (and many others, irrelevant to the merge) I want the dataset AB with all these columns Open from A - a difftime (time of day) Time from B - a difftime (time of day) Name (same in A & B) - a factor, does NOT index rows, i.e., there are _many_ rows in both A & B with

Hi All, I got a doubt about something I want to do with asterisk. I have this office (site a) with only a Panasonic analog PBX and another office (site b) with an Asterisk Box with an ADIT 600 . I want to interconnect both via IAX. Is it possible to put a new asterisk box in site a without the channel bank and put a card (FXS or FXO???) and connect it to the pbx as a CO line ? What

I've seen mention of it here and there... does anyone have mgcp working with a 7960? I've gotten the phone to work in "basic phone mode", is that all I'll get, or am I missing something? ___________________________________________________________ Steve Creel screel@turbs.com

I've been trying to find a way to connect incoming calls to my cell phone when I'm not in the office. I would like to have asterisk call the cell phone (or any other phone for that matter), and provide me the option to connect to the call. I figure I could park the call, use /var/spool/asterisk/outgoing/ to generate a call to the cell phone and put it into a context somewhere. Now

It just dawned on me as I was playing with the manager interface - it can't be very difficult at all to write an Win32 app that serves as a "lamp field". Between 'Newchannel', 'Newstate', and 'Hangup' events, all of the information is there. I've heard several requests for line appearances, but mgcp and sccp channels don't currently include support.

How can I loop through something x number of times in the dialplan? i.e. if I get an invalid extension I want to re-play the menu, but not forever. Maybe 3 tries or something. I'm pretty sure that I've seen it before, where you can increment a variable and do "Gotos" based on it. But I've searched the Asterisk handbook, searched the user archives, and Googled for it,

I bought a couple of 7905G phones with a Callmanager license but i found on the site these phones can have a SIP image (which i downloaded) but before i upload the image i want to know if anybody tested them ? Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/4c2821af/attachment.htm

Does anyone have any suggestions for excluding certain users from the directory? Can I just leave the 'Name' field empty in voicemail.conf Certain voicemail boxes shouldn't show up in the directory (company president, etc). I assume this can be handled safely by just leaving out the 'name' in voicemail.conf To go a step further, it would be good to allow them to put the