similar to: chanh323 dialling

I have just compiled & installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's'

Hi everybody, I'm trying to use msn with * and for that, I'm reading all information on the mailing list. You used to recommend the page http://mcleod.pbx.nq.net/msn/, but I always get an error while opening. Has it changed? Is there another one? Thanks cmayor ___________________________________________________ Yahoo! Messenger - Nueva versi?n GRATIS Super Webcam, voz, caritas animadas, y

Hi, Talking about messenger,,, it's still necesary to do HKEY_CURRENT_USER\Software\Microsoft\MessengerService\Corp2PC_Phone equals to '1' ??? But it's still sending the '+' digit, so it's necesary to stripMSD? Thanks a lot cmayor ___________________________________________________ Yahoo! Messenger - Nueva versi?n GRATIS Super Webcam, voz, caritas animadas, y m?s...

Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I asked a time ago what were callgroup and pickup group used for. I have done some proofs and all, and I'm not sure if I have pick the idea up well!! That's what I understand: For example: group=1 callgroup =2 and pickupgroup=2 and my phone is a membership of the group 1. that's mean that when a phone that belong to group 2 is ringing, I'll be able to answer this call dialing

Hi since "callerid=" in sip.conf doesn't set the Caller ID, I suppose it must be there for some other reason. Is this a not-yet-working feature for future releases of Asterisk? If not, what does it actually do? thanks regards bk

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Hi! Is it possible to merge two ext3/ext2 partitions into ONE ext3/ext2 partition? -- Ralf Hildebrandt (Im Auftrag des Referat V a) Ralf.Hildebrandt at charite.de Charite - Universit?tsmedizin Berlin Tel. +49 (0)30-450 570-155 Gemeinsame Einrichtung von FU- und HU-Berlin Fax. +49 (0)30-450 570-916 IT-Zentrum Standort Campus Mitte AIM. ralfpostfix

just three "stupid" questions I need to ask .. 1. what's the sequence to press on a SIP phone to transfer a call to another extension. 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? 3. what's the extensions.conf syntax to dial two SIP extensions at once? many thanks Dave

This is what happens when i run the ./tools/wineinstall WINE Installer v0.74 Running configure... configure: creating cache config.cache checking build system type... i686-pc-linux-gnuoldld checking host system type... i686-pc-linux-gnuoldld checking whether make sets $(MAKE)... yes checking for gcc... no checking for cc... no checking for cc... no checking for cl... no configure: error: no

hi .. i require to interface asterisk to a 60 line analog PBX in a hotel. I was thinking of giving Asterisk a couple of PBX lines interfaced through cards, and then place outgoing calls through SIP/H323 and a DSL connection. analog extension lines <--> analog pbx <-->asterisk <--> SIP --> termination I do not need incoming calls to the lines. My question is this : if I

i've just lost $2000 dollars or so on my first commercial asterisk installation .. i'm running a PIV class server, three Digium Wildcard FXO cards, and 10 Grandstream Budgettone SIP phones. The system was to be a PBX for a small company. After over 2 months of pissing about, the client has had his fill of asterisk problems, and asked me to take my equipment out of the building. Obviously,

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the

I have a test SIP account set up with WorldCom and I have been trying to have Asterisk register to the WorldCom server with no luck. It appears that the SIP headers are different coming from Asterisk. I have included a packet capture from a successful login with a Windows Messenger client for reference. I have also copied in the SIP packet I captured with sip debug turned on. In my sip.conf file,

Hi everybody and sorry for posting this again to the list. I don't want you guys to think that I'm DEMANDING FOR SUPPORT (we all have just had several discusions about that) but my experience tell me that if a posted question is easy enough, it is answered immediatly, or it will never be answerd!!!(people forget it...) Mine are two very simple questions about modules.conf. It's only a

Greetings to one and all on this fine list; We have the current system: Meridian 1 Option 11 +-------------------+ | | ISDN/30 (DASS/2) ===> |NTAK79BB (2MB Pri) | | |<-->4x16 port Digital / 1x16 port Analogue ISDN/30 (EUROIDSN) ===> |NTBK50AA (2MB Pri)

I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan

hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the call and hang up .. so far i've got as far as a blind transfer by pressing transfer button and

hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3

Hi, I am trying to install passenger 3.0 for apache2 on Ubuntu 10.10. It says that I need to install libopenssl-ruby. But when I did ''sudo apt-get install libopenssl-ruby'', I got the following error. Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting ''libruby'' instead of