search for: herchi

Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help,

Hello everybody, I?ve been pulling my hair for a week now over a problem, and I really don?t know where to look anymore. Here?s my setup: There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I can use it to send and receive calls from physical phones attached to it. I have setup Asterisk 1.0.3, with H323 and openH323, and on the same server I also setup GnuGK (10.253.30.1). I

...Executing StripMSD("H323/ip$10.253.30.11:1119/284", "3") in new stack -- Executing Goto("H323/ip$10.253.30.11:1119/284", "SIP|6666|1") in new stack -- Goto (SIP,6666,1) -- Executing Dial("H323/ip$10.253.30.11:1119/284", "SIP/silviu.herchi") in new stack -- Called silviu.herchi -- SIP/silviu.herchi-b027 is ringing Sending alerting -- Received Facility message... -- Received Facility message... -- Received Facility message... -- Received Facility message... -- Received F...

Hi all, I've been pulling my hair out for two days over this problem... I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem

Hi all, I have thoroughly read the available documentation and I can't seem to find a workaround for my setup... I'm trying to create a phone conference line that users would call using a unique phone number (no matter if they are moderators or just plain users). I use Asterisk 1.2.6 The available conferences are defined as follows: conf => 1000,user pin1, moderator pin1 conf =>

Hello, Here's what I'd like to do: call my Asterisk box from a phone, hangup after a few rings, then Asterisk calls me back and presents a dialtone, than I can dial any valid number in the context the call originated. I've done it with CAPI (thanks to the script on http://www.junghanns.net/asterisk/page14.html), I'd like to do it with H323. Problem is, how to present a

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf

Hi everybody, Recently my Asterisk server started to behave strangely: in some cases (hard to diagnose and reproduce), the SIP module stops responding, and the log is filled with messages "Failed to grab lock, trying again..." (about a hundred messages or so per second). This often happens at the end of a SIP call (right after "== Spawn extension (SipExtension, 029156044, 1)

Hi everybody, Recently my Asterisk server started to behave strangely: in some cases (hard to diagnose and reproduce), the SIP module stops responding, and the log is filled with messages "Failed to grab lock, trying again..." (about a hundred messages or so per second). This often happens at the end of a SIP call (right after "== Spawn extension (SipExtension, 029156044, 1)

Hello, I think you should handle the fax in the h (for Hangup) extension (which is, after your fax was received), instead of using the priorities following the fax reception (as in your example). Have a look at the different examples in the wiki, like http://www.voip-info.org/wiki-Asterisk+fax: [fax] exten => 666,1,Macro(faxreceive) exten => h,1,system(/usr/sbin/mailfax ${FAXFILE}