similar to: Upgrade to v.1.2.31 ... weird change

Displaying 20 results from an estimated 1000 matches similar to: "Upgrade to v.1.2.31 ... weird change"

2009 Jun 09
5
IAX2 issue?
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
2006 Jun 22
1
South Africa DIDs
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
2007 Apr 30
1
IVR dictionary dial-plan
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"
2006 Sep 19
1
Mounting home directories on NAS
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 =
2006 Mar 14
3
EICON Diva 4BRI
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
2006 Feb 17
3
g.729 woes
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
2006 Feb 17
0
codec negotiation with SPA-3K
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
2006 Mar 26
0
UK EI
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
2006 Apr 10
1
SIP channel unavailable/busy/really not there
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
2006 Apr 21
1
Definitive list of sounds
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
2006 Jun 06
0
[asterisk-dev] UK Male English Voices
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
2007 Jan 02
0
[asterisk-biz] Slightly updated UK English voice prompts
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
2007 Mar 27
1
UK BT PRI
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
2007 Apr 21
1
UK zaptel and zapata.conf for TDM400P
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
2007 May 18
2
zap fallback
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
2007 Jun 06
2
shorting flash time
Is there anyway to change the "flash" time on a TDM400 phone port (a user has a phone that seems to generate a short flash which isn't being picked up). 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
2006 Jun 06
2
UK Male English Voices
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
2006 Apr 19
3
SLIN format
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
2007 Apr 25
3
FYI
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
2006 Mar 13
4
priorityjumping=no
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 =>