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...---------+ | | ISDN/30 (DASS/2) ===> |NTAK79BB (2MB Pri) | | |<-->4x16 port Digital / 1x16 port Analogue ISDN/30 (EUROIDSN) ===> |NTBK50AA (2MB Pri) | | | Melita predictive Dialler | NTAK79BC (2MB Pri)|<===> +---------------------------+ | | | | | NTAK79BC (2MB Pri)|<===> | Aculab E1 SC-BUS ISA | | | | (D...

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 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

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

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

what is the format for an h323 entry in the dialplan? can I use chan_h323 without compiling anything else or should I compile oh323? basically what's the best way :) cheers Dave

...subnet REMOTE_BROADCAST_SUBNET - name not found. The other IP address is 192.68.22.214 Unfortunately, the log does not have time stamps, so I don't know when this is happening. Neither of these networks are mine (an nslookup shows lbb.ofthe.net - a local Internet Service Provider and host214.melita.com a company in Georgia - we are in Texas). So, my question - what EXACTLY does this message mean? No, I don't have ports 137 and 139 blocked, but I supposed I should... === Tim --------------------------------------------------------------------- | Tim Winders, CNE, MCP | Email:...

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'

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,

...d in the OH323.conf file and it seems to be using it .. connection is not established, I have pasted a dump file below .. anyone knows what's wrong ? i'm beyond my level of asterisk knowledge at this point :( thanks Dave ----- Original Message ----- From: "root" <root@soyuz.melita.net> To: <david@melita.net> Sent: Thursday, July 10, 2003 10:11 PM > 0:00.006 OpenH323 Wrapper OpenH323 Wrapper Version 0.0alpha0 by inAccess Networks (www.inaccessnetworks.com) on Unix Linux (2.4.20-8-i686) at 2003/7/10 22:10:37.181 > 0:00.008 OpenH323 Wrapper H3...

hi there, I have just downloaded and installed asterisk a couple of days ago, it compiled correctly and starts up and runs, on a Redhat 9 system freshly installed for testing. I don't have any extra hardware installed so far, was attempting to just try out connectivity. I am having some probs with the configuration, maybe someone out there can give me some tips : firstly on modifying the

I am trying to get asterisk to dial this address : sip:723@216.52.153.207 Using a softphone on my PC (217.168.168.49) it dials immediately and I get a voice prompt .. I have configured an extension, 1303 on asterisk, modifying the demo configuration : exten => 1303,1,Dial(SIP/723@216.52.153.207) When from my softphone I dial sip:1303@217.168.168.51 on the console I get : -- Executing

I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing

hi, does anyone have a problem with OH323 crashing with a segmentation fault whenever anything tries to connect to it ??? are the current CVS versions OK? Would like to speak to someone with a bit of OH323 experience, so if u're in a good mood to help, please do :) cheers Dave

not really asterisk related this, but is it normal for a mail to take so long to be resent through the mailing list server? i'm speaking about 20 minute + delays here .. (or it it only me ?) cheers Dave

My system is an asterisk machine, with an E1 card (functioning) and forwarding calls to a remote SIP address .. when a call connects I am getting the following error : NOTICE[1240577216]: File rtp.c, Line 330 (ast_rtp_read): Unknown RTP codec 19 received can anybody tell me what this means (& how I may fix it ?) cheers Dave ps. the service i'm connecting to uses G723

hi .. maybe someone can help me, I seem to have lost the sheet of paper that comes with an E100P card and tells you how to compile the stuff it requires to run. I'm trying to move my Asterisk to a different box and at this time totally stuck. Could someone be kind enough as to mail me a PDF of it ?? many thanks Dave

hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get "Invalid CSeq Number" back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave

hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get "Invalid CSeq Number" back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave

i'm getting Asterisk to dial an h323 call termination service .. right now getting this message: -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial("Zap/1-1", "OH323/h323:723@216.52.153.206") in new stack 5:59.330 H323 Cleaner H323