Re: [Asterisk-Users] Menu/IVR and transfering to an extension after pressing an option number.
Try:
; Dial an extension
exten = _1.,1,Dial(SIP/${EXTEN:1})
That would match anything entered starting with a 1 and dial it without the
first digit.
Jeremy Koski wrote:
Here's what I'm trying to accomplish:
Press 1 to transfer to extension
Press 2 for Directory
Press 0 for Operator
Got directory and operator working. My problem is with transfering to an
extension after pressing 1. Asterisk keeps adding the 1 to the extension
that I need to transfer to (extension 500 would be 1500 to asterisk
rathern than 500). I've tried ${EXTEN}, ${EXTEN:1},2,3,4 etc..No luck.
This is what I have currently:
[MainMenu]
include = extensions
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,Set(TIMEOUT(digit)=5)
exten = s,4,Set(TIMEOUT(response)=10)
exten = s,5,Background(danner) ; Play Welcome to companyname greeting
exten = i,1,Playback(invalid)
exten = i,2,GoTo(MainMenu,s,1)
; Dial an extension
exten = 1,1,Dial(SIP/${EXTEN:1})
; Go to the sales department
exten = 2,1,Directory(extensions) ;
; Leave; a voicemail for the operator
exten = 0,1,Dial(${P100}${P103},30,t) ;
Can anybody be of any assistance?
Thanks.
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Re: [Asterisk-Users] why won;t my voice files play?
Asterisk only needs a cLock to play MOH afaik, on 2.6.x kernels you don't need any timing help, on 2.4.x you can use ztdummy on the USB drivers Damon Estep wrote: Sounds like you lost timing, either because a zaptel device driver did not load or ztdummy did not load if you have no zap hardware. * needs a cock a what ? :P to play sounds and keep timing, the clock comes from the PSTN, zap hardware, or ztdummy depending on how you are set up. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Wednesday, August 31, 2005 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] why won;t my voice files play? I just recompiled my version from this morning's CVS Head. My systems voice files (voicemail, time etc) were playing nicely. Until that is I added an extension and now the files won't play. Worse than that, * thinks the files have played and goes to the next step in the dial plan. What gives? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- InfoPact Netwerkdiensten B.V. http://www.infopact.nl/ Emmastraat 11-13 3255 BD Oude Tonge tel. +31 (0)187 64 77 11 mob. +31 (0)645 18 69 67 fax. +31 (0)187 64 77 99 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why won;t my voice files play?
Here, have a couple of my s ;) Steve Langstaff wrote: Here, have an 'l' - I've go a couple spare on my keyboard :) I guess it needs a clock to play sounds... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Damon Estep Sent: 31 August 2005 14:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] why won;t my voice files play? Sounds like you lost timing, either because a zaptel device driver did not load or ztdummy did not load if you have no zap hardware. * needs a cock to play sounds and keep timing, the clock comes from the PSTN, zap hardware, or ztdummy depending on how you are set up. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- InfoPact Netwerkdiensten B.V. http://www.infopact.nl/ Emmastraat 11-13 3255 BD Oude Tonge tel. +31 (0)187 64 77 11 mob. +31 (0)645 18 69 67 fax. +31 (0)187 64 77 99 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
That should be controllable by a weight, for example 2 peers: A -- G729, G711 B -- G711, G729 What's currently happening is that * starts transcoding between the two (g729 for A and G711 for B), what i would like is to apply a weight to peer A so that the codec of choise at both sides becomes the preffered choise of A (G729) on both sides so there won't be any transcoding. This would allow for some nice things as fax passtrough (A and B has to use G711 then, but if the weigted A says G711, B would use G711 to). Kind regards, Erik Brian West wrote: Here is an example: Call comes in via PSTN... ulaw is the native format of the channel. On the sip side you have g729,ulaw as the codec order. That call will end up being ulaw because we send the native format as our first choice above all because we don't want to transcode. /b On Aug 15, 2005, at 1:10 PM, Tony Hoyle wrote: Pavel Jezek wrote: Hi, asterisk will negotiate codecs for both parties independently (use sip show peer peer and look for codec order entry), so, if you have prefered codec g729 for your sip phone/peer, asterisk will use them (regardles of codec setting for other party - if codecs does not match, asterisk will try to transcode between) imho ;-) It does seem to be a weakness of asterisk.. it's creating load on the server when it doesn't need to. Really it should look at the capabilities of both ends and see if there's a common set, and only start transcoding if there's no overlap. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- InfoPact Netwerkdiensten B.V. http://www.infopact.nl/ Emmastraat 11-13 3255 BD Oude Tonge tel. +31 (0)187 64 77 11 mob. +31 (0)645 18 69 67 fax. +31 (0)187 64 77 99 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79XX and VLANS
If you're using Cisco Switches: Logon to the switch and go to config mode int fa0/1 switchport access voice vlan untagged sometheing in that direction configures the CDP to set the phone to untagged frames. Julian Lyndon-Smith wrote: How ? Where ? I've been wanting to do this for ages, and never found an option to do so ! Please Please Please tell all. (I hate begging, but sometimes ) Julian. Eric Wieling aka ManxPower wrote: Matthew Boehm wrote: Hey gang, We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We are also using all Cisco Switches and Routers. Everything works great except that when you reboot a phone it takes like 3-5 minutes for it to come up. The phones spend tons of time 'Configuring VLAN..' We don't run any VLANs. Is there some way to skip this? In the 'Network Settings' I have both 'Operational VLAN Id' and 'Admin VLAN Id' set to blank values. Disable CDP on the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transcoding
Rich Adamson wrote: I though that Asterisk would transcode between codecs! All my SIP devices support G729a 711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept a call from a SIP device using G729a and then complains that it can't translate into G711 to go onto the ISDN network. Does anybody know if there is some setting somewhere or if this is how it is supposed to work What does the sip debug show? Any CLI data to give us a clue? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users And you did buy some G729 codec licences? * won't do anything but passtrough without them ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENTt: GPL Asterisk Billing Software
What happens if the rate changes mid call? IE, call starts @ 18.30 and lasts till 19.15 Rate changes @1900 to off-peak. Darren Wiebe wrote: Partially. I have not finished the script that will limit the calls depending on the money available. Darren Wiebe [EMAIL PROTECTED] VoIP Newbie wrote: Does it support pre-paid billing? On 5/30/05, Darren Wiebe [EMAIL PROTECTED] wrote: El Flynn wrote: Darren Wiebe wrote: Good Day, I'm finally getting around to officially announcing ASTPP. For the last 6 months, I've been working on converting ASTCC into a decent billing package for asterisk. The link in the original email opens a page that says Download the latest version of the code from http://www.aleph-com.net/astpp.html Has anyone else been able to download this code? I can't seem to find a link on their site to the code itself, and the astpp.html page brings up a Not Found... Sorry, I missed that old link. I just got everything moved onto the wiki on Friday night. Please download the code off of the cvs server. I'm getting close to ready to release version 1.0 and then I will post a copy on the website. At present, I believe the only show stopping bug is in the AgileBill integration. That will be fixed shortly. Darren Wiebe [EMAIL PROTECTED] Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- InfoPact Netwerkdiensten B.V. http://www.infopact.nl/ Emmastraat 11-13 3255 BD Oude Tonge tel. +31 (0)187 64 77 11 mob. +31 (0)645 18 69 67 fax. +31 (0)187 64 77 99 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk not recognising On Hold
I'm having some troubles with my * machine, when i place a call on hold the callee doesn't hear any MOH and the call is dropped because of lack of RTP. I also don't see * starting MOH on the SIP channel the callee is on (moh class is defined, there are MP3 files and mpg123 is active). I'm using * 1.0.6 right now with Cisco 7940's, i can see * recieving a SIP invite with c=0.0.0.0 so that should work, i can allso see the invite back to the phone when getting the call out of hold. Because of this problem attented transfers won't work correct either (since the other side of the call gets dropped before the call is transferred). All calls are SIP--SIP. Any ideas? Kind regards, E. Versaevel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip OPTIONS
That would be nice, SER does have the possibility to answer an OPTIONS correctly, but * indeed answers with a 404, i'm now using sipsak to register a test user, that also works. Andres wrote: Erik Versaevel wrote: Hello all, Can asterisk itself respond to a SIP OPTIONS packet? i wan't to check if asterisk is still alive by using sipsak (because of nagois mon) Sure it does. It answers with 404 Not Found. We monitor all our SER and Asterisk servers with a Nagios plugin based on sipsak that was witten by somebody on the SER list. We just modified it a bit to fit Asterisk. If you are interested in the plugin I can post it tomorrow. Andres Network Admin http://www.telesip.net Kind regards, E. Versaevel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip OPTIONS
Hello all, Can asterisk itself respond to a SIP OPTIONS packet? i wan't to check if asterisk is still alive by using sipsak (because of nagois mon) Kind regards, E. Versaevel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Displaying incoming e.164 callers number - how?
Mark Elkins wrote:
On Tue, 2005-01-04 at 15:45 +0200, Mark Elkins wrote:
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
Playing with myself again - that is - I called myself - and
got the caller ID of '27128070590'... not quite what I wanted...
In my extensions - I have...
[fromaix]
exten = 27128070590,1,Goto(default,s,1)
And again - changed the above to...
[fromaix]
exten = 27128070590,1,GotoIf($[${CALLERIDNUM:0:2} = 27]?2:4)
exten = 27128070590,2,SetGlobalVar(CALLERIDNUM=0${CALLERIDNUM:2})
exten = 27128070590,3,Goto(default,s,1)
exten = 27128070590,4,SetGlobalVar(CALLERIDNUM=09${CALLERIDNUM})
exten = 27128070590,5,Goto(default,s,1)
Default section looks like...
[default] ; what people will get when they call me.
exten = s,1,NoOp(CALLER=${CALLERIDNUM})
exten = s,2,Answer()
Logic flow is meant to be..
1 - if CallerIDNum starts with '27' - goto line 2 - else goto line 4
2 - Remove the first two digits off the CallerIDNum, replace with '0'
3 - Goto my default section - normal processing
4 - Prepend the CallerIDNum with '09'
5 - Goto my default section - normal processing
Problems -
The callerIDNum variable does not change :-( I thought that is what
'SetGlobalVar' was ment to do??? Seems to be ReadOnly or in a local
context... - how do I 'export' the change?
The Console shows...
-- Executing GotoIf(IAX2/[EMAIL PROTECTED]:4569/4, 1?2:4) in
new stack
-- Goto (fromaix,27128070590,2)
-- Executing SetGlobalVar(IAX2/[EMAIL PROTECTED]:4569/4,
CALLERIDNUM=0128070590) in new stack
-- Setting global variable 'CALLERIDNUM' to '0128070590'
-- Executing Goto(IAX2/[EMAIL PROTECTED]:4569/4, default|s|1)
in new stack
-- Goto (default,s,1)
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569/4,
CALLER=27128070590) in new stack
how about SetCallerId(12345) ;)
ie
exten= 27128070590, 2, setcallerid(0${CALLERIDNUM});
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID
SetCallerID
Synopsis
Set CallerID
Description
SetCallerID(clid[|a])
Set Caller*ID on a call to a new value. Sets ANI wiki-ANI as well if a
flag is used. Note that the variable wiki-Asterisk+variables
*${CALLERID}* contains the current call's Caller ID (name and number).
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Re: [Asterisk-Users] codec preferences
Sorry for my hars anwer :) You are able to call into asterisk (moh for example) using G729? Roy Sigurd Karlsbakk wrote: I did read the docs and I did purchase the G.729 license. sipgw1*CLI show g729 0/0 encoders/decoders of 25 licensed channels are currently in use roy On Dec 28, 2004, at 9:31, E. Versaevel wrote: You did read the documentation I presume? It clearly states that you need to have separate G729 licenses to have * talk G729, without them * can only pass thru G729, not transcode it (Zap - G729). Since both your sip phones do G729 * is just performing pass thru. Kind regards, E. Versaevel -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Roy Sigurd Karlsbakk Verzonden: maandag 27 december 2004 21:07 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] codec preferences hi Username : 112 Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (gsm|g729|g726|alaw|ulaw) the above is from SIP SHOW PEER 112, and as it clearly shows, g.729 is preferred before alaw. If I dial this SIP - * - SIP from a phone with G.729 enabled, it uses G.729. However, if I dial from my cell phone - GSM - PSTN - * - SIP, the call uses ALAW, which I thought it shouldn't. I'm using Asterisk CVS-v1-0-12/16/04-04:15:36 with the patch from http://bugs.digium.com/bug_view_page.php?bug_id=0003106 all comments welcome roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
