similar to: g.729 woes

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 =>

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

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"

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

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 =

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

Just founnd a weirdy. My end is Asterisk 1.2.32 using an IAX2 link to the US. The IP address of the remote end changed (though in the config file it's registered as a name i.e. asterisk.remote.end), my system didn't recognised the IP change, it must be cached once and then the cached value used for ever. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857

Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100802/4f58fea7/attachment.htm

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Hello all, I really appreciate the replies I've gotten about this so far (especially the support for wanting to run it on Solaris!). The core issue seems to have been missed, though -- is there any way to run a complete Asterisk solution on Solaris 10 (including music-on-hold and conferencing)? This probably comes down to a few issues: - Is ztdummy (a component of Zaptel) *really*

Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. Thanks, James

Has any one found any issues (bug?) with ver 1.2.5 as CallerID generated on an outgoing FXS port (to the handset) fails when UK tones are used, with a message 'Didn't finish Caller-ID spill. Cancelling.' Any tips on getting this running ? Looked at Mantis, but only known bugs seem to relate to XP100 cards - not the TDM card. Thanks

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