similar to: shorting flash time

Is it possible to get Joburg DIDs (probably need 4 at the moment), to be delivered via SIP preferrably to UK. If it's legal, please send pricing. Thanks Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Does anyone know of an (E)AGI or program to develop a IVR dial-plan which will take a list of words and then do something when a unique branch has been found. i.e. Say there's 3 words demon deacon bishop On a phone they'd be represented as 33666 332266 247467 So if the user enters "2" we know they want bishop if they enter "336" they want demon and "332"

I'm using Samba 2.2.12 (Sun Solaris 9 version) and it's all working fine as a PDC for a small domain. I would like to use the /home construct so that the home directories are now mounted from a NAS rather than the local box. In smb.conf I have ; User Profiles logon path = \\%N\profiles\%U ; Where is a user's home directory and should it be mounted logon drive = Z: logon home =

I'm having trouble with Asterisk-1.2.4 negotiating codecs with a Sipura 3000 which is running the latest v3 firmware. The SPA-3K seems to use the "preferred" codec only and doesn't negotiate? The SPA is set to no in "use only preferred codec". Does anyone know if Sipura will support gsm at some point? I this a bug with the SPA or codec negotiation stuff? Thanks

I'm using a Digium TE411P connected to a UK switch (EuroISDN). Everything is working, but if I dial a busy number (from SIP) is seems to stay busy until I hang up, even though the dial-plan drops through some other stuff using CALLSTATUS variable (i.e. S-BUSY), none of the timeouts come into play. Any ideas? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US

Is there a way to differentiate between a SIP address which hasn't registered (but is within sip.conf) and one that's not there at all (i.e. not in sip.conf) using a straight dialplan. I'd like to differentiate actions depending the state of a SIP device and whether it's in my config or not (if that makes sense, basic automap of dial-in lines to sip phones, but if they've

Is there a list of sounds (base - as with Asterisk itself, and additional) for the 1.2 release. As in a list with what the content of each file is. There's a list for 1.0.7 on the wiki, but that seems woefully out of date. Any help appreciated. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo

I'd like to announce that the UK Male English Voices are now up on http://www.tel.net/ There's a complete set of base sounds and additional sounds (it should be complete compared to current Asterisk and Asterisk-Sounds-1.2.1). There's also a set with the word 'pound' replaced by 'hash' for both the base and additional sounds (only the actual replacements not a

I believe there were some new prompts added for 1.4 for Directory Info. These have now been added to http://www.tel.net Have a good 2007. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Has anyone got a working zaptel.conf and zapata.conf for a Digium Wildcard TE110P T1/E1 Card. It's connected to a BT ISDN PRI (EuroISDN) with 24 channels. Inbound works fine, but outbound isn't setting CLI (it seems the line supports 6 digit CLI). Inbound CLI works fine. In the dial-plan using Set(CALLERID(num)=123456) then Dial(Zap/g1/01234567||frT) Where 123456 is in the range of BT

Has anyone got a sensible zaptel.conf and zapata.conf for 2 TDM400P's working with UK set-up. They're set-up with 7 analogue phones and 1 PSTN port. Currently zaptel.conf has fxoks=1-7 fxsks=8 loadzone=uk defaultzone=uk It's really zapata.conf that would be useful. Currently using the zaptel/asterisk that comes with Ubuntu (latest) which needed a bit of tweaking (1.2.16), but could

I'm trying to get zap fallback to VoIP working. I dial the zap channel and if it fails I want to then try another route. If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if there's no dial-tone, it doesn't seem to detect this? Using a TDM400 with UK settings. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20

I'd like to announce that the UK Male English Voices are now up on http://www.tel.net/ There's a complete set of base sounds and additional sounds (it should be complete compared to current Asterisk and Asterisk-Sounds-1.2.1). There's also a set with the word 'pound' replaced by 'hash' for both the base and additional sounds (only the actual replacements not a

In sox terms is SLIN .ul (as in unsigned linear). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Are there any step by step instrunctions on how to install drivers and I guess bristuff for this card? Just need to use it to handle voice on 2 BRI circuits (UK) then utilise with Asterisk and some Digium cards handling POTS phones (and some VoIP out the back). It's the EICON card stuff and how to make it all work I'm finding confusing? Steve -- NetTek Ltd UK mob +44-(0)7775 755503

I have some Digium licensed Digium codecs, but when making a call and transcoding the call is only heard in one direction? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Just been getting lots of failed SIP registrations to a system here. All coming from Taiwan. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

I've been trying to use a set-up whereby I have several TA's connected to an Asterisk server (1.2.4) and they act like they're in a hunt-group i.e. try the first, if busy jump to the next etc. in my extensions.conf I had something like [inbound-trunk] exten => 441234123456,1,Dial(SIP/s1a,20,r) exten => 441234123456,102,Dial(SIP/s2a,20,r) exten =>

I think it happened when I upgraded an install to 1.2.31 The variable CALLERIDNUM no longer works and CallerID(num) has to be used. I know the initial one was being depreciated, but I didn't see any mention of it. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk

Hello. I'm trying to transfer calls from an analog phone (Zap/1, TDM400P card) to some other terminal connected to my Asterisk PBX. If I make a flash hook pressing the phone hangup button quickly it works as expected, I get a new dialtone and the other side is put on hold. But I would like to use my phone's "R" key instead for some different reasons (it's quite easier to use