[Asterisk-Users] app_dbodbc for asterisk stable 1.09
Hi, Has anyone manage to comile app_dbodbc or ast_data with the latest stable release (1.09). If so can you give some guidence on howto do it as I have trouble getting either working. Umar ___ 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 Manager interface, setting global vars
Hi all, Does anyone know of a way to setup global var using the manager interface. Basically I want to be able to have multiple manager clients login, however in a sort of master slave scenario. So the first client that logs in, sets a global variable which tells subsequent clients at least one client is already logged in. The Master would then set additional variables which the slaves would periodically read. Is this possible ? Thanks in advance for any help. Umar ___ 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] Can't get format_mp3 to work for music on hold
Anymore ideas ? anyone ? On Sun, 27 Mar 2005 23:47:50 +0100, Umar Sear [EMAIL PROTECTED] wrote: I removed mpg123 as per instruction on the wiki as I am trying to use native mp3 (format_mp3) Re-installing it makes no difference. Thanks Umar On Sun, 27 Mar 2005 12:24:32 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Sun, 2005-03-27 at 13:41 +0100, Umar Sear wrote: Hi Guys, I am having trouble trying to get format_mp3 working to play music on hold. I have followed the instructions in the read-me and the wiki however it seems after un-installing mpg123, asterisk is not even attempting to play MOH. Executing Answer(SIP/-56f4, ) in new stack -- Executing MusicOnHold(SIP/-56f4, default) in new stack Mar 27 13:40:05 WARNING[13181]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class 'default') on channel SIP/-56f4 == Spawn extension (default, , 2) exited non-zero on 'SIP Please tell me I am doing something stupid ! as I would love to know, before I pull all my hair out :-) It can't be installed because you have uninstalled mpg123. Why did you uninstall mpg123? Reinstall mpg123 and stop messing with your hair. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Can't get format_mp3 to work for music on hold
I downloaded and installed the cvs version on my test machine. It works with the exact configuration that I am using with the 1.x stable version. So I assume that format_mp3 does not work with version 1.x, please correct me if I am wrong. Umar On Wed, 30 Mar 2005 20:08:04 +0100, Umar Sear [EMAIL PROTECTED] wrote: Anymore ideas ? anyone ? On Sun, 27 Mar 2005 23:47:50 +0100, Umar Sear [EMAIL PROTECTED] wrote: I removed mpg123 as per instruction on the wiki as I am trying to use native mp3 (format_mp3) Re-installing it makes no difference. Thanks Umar On Sun, 27 Mar 2005 12:24:32 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Sun, 2005-03-27 at 13:41 +0100, Umar Sear wrote: Hi Guys, I am having trouble trying to get format_mp3 working to play music on hold. I have followed the instructions in the read-me and the wiki however it seems after un-installing mpg123, asterisk is not even attempting to play MOH. Executing Answer(SIP/-56f4, ) in new stack -- Executing MusicOnHold(SIP/-56f4, default) in new stack Mar 27 13:40:05 WARNING[13181]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class 'default') on channel SIP/-56f4 == Spawn extension (default, , 2) exited non-zero on 'SIP Please tell me I am doing something stupid ! as I would love to know, before I pull all my hair out :-) It can't be installed because you have uninstalled mpg123. Why did you uninstall mpg123? Reinstall mpg123 and stop messing with your hair. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] ACD queue question
are you restarting asterisk or reloading after changing you configuration. Umar On Wed, 30 Mar 2005 19:33:42 -0600, Eric Rees [EMAIL PROTECTED] wrote: I tried leastrecent. I did change the strategy, but didn't make a difference. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: Wednesday, March 30, 2005 6:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] ACD queue question Using which strategy? Remember, if you change strategies and reload, it'll forget where it was and start over. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees Sent: Wednesday, March 30, 2005 6:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ACD queue question That's what I thought would happen, but after about an hour and 100 or so incoming calls, it was still ringing the agents in the order that they were listed in the agents.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: Tuesday, March 29, 2005 10:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] ACD queue question The first call for each agent probably goes that way, but then after a few calls have rolled through the queue, the strategy you specify (like LeastRecent) should come into play. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees Sent: Tuesday, March 29, 2005 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ACD queue question I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong. ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 ___ 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
Re: [Asterisk-Users] cmd Authenticiation
How would you distinguish different users ? Umar On Thu, 31 Mar 2005 11:46:13 +0800, Simon [EMAIL PROTECTED] wrote: Hi folks, Sorry to post a simple command, I am deep into this and hope any help from the experts. I am using the command Authenticate as explained in wi-ki: I am managed to authenticiate with a single global password but my requirement will every user have their own password and contexts to call Please help me Thank you Simon ___ 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] Question
Which is not needed anymore as app_voicemail will create a mailbox automatically. On Tue, 29 Mar 2005 18:21:03 +0200, Guy Decarpentrie [EMAIL PROTECTED] wrote: Le mardi 29 Mars 2005 18:13, Parker, Blake (MIS) a écrit : What is the command to create a new voicemail box? addmailbox in /asterisk_directory/contrib/scripts Blake ___ 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] Can't get format_mp3 to work for music on hold
Hi Guys, I am having trouble trying to get format_mp3 working to play music on hold. I have followed the instructions in the read-me and the wiki however it seems after un-installing mpg123, asterisk is not even attempting to play MOH. My musiconhold.conf is ; Music on hold class definitions ; [classes] [moh_files] default = /var/lib/asterisk/moh-native ;default = quietmp3:/var/lib/asterisk/mohmp3 ;loud = mp3:/var/lib/asterisk/mohmp3 ;random = quietmp3:/var/lib/asterisk/mohmp3,-z ;unbuffered = mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf = quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters (specifically embedded spaces) ;manual = custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s When I call my queue I get this on the console Connected to Asterisk 1.0.7 currently running on aovas1 (pid = 13181) Verbosity is at least 3 -- Remote UNIX connection -- Executing Answer(SIP/08452418339-556a, ) in new stack -- Executing Queue(SIP/08452418339-556a, testq) in new stack -- Playing 'queue-youarenext' (language 'en') -- Told SIP/08452418339-556a in testq their queue position (which was 1) -- Playing 'queue-thankyou' (language 'en') -- Playing 'queue-youarenext' (language 'en') -- Told SIP/08452418339-556a in testq their queue position (which was 1) -- Playing 'queue-thankyou' (language 'en') The extensions.conf is configured like ... exten = X,1,Answer exten = X,2,Queue(testq) Where is obviously the number I am dialling. I have also tried to following extension with no joy exten = ,1,Answer exten = ,2,MusicOnHold(default) exten = ,3,Busy In which case I get this on the cli Executing Answer(SIP/-56f4, ) in new stack -- Executing MusicOnHold(SIP/-56f4, default) in new stack Mar 27 13:40:05 WARNING[13181]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class 'default') on channel SIP/-56f4 == Spawn extension (default, , 2) exited non-zero on 'SIP Please tell me I am doing something stupid ! as I would love to know, before I pull all my hair out :-) Umar ___ 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] Can't get format_mp3 to work for music on hold
On Sun, 27 Mar 2005 14:56:12 +0200, Martijn van Oosterhout [EMAIL PROTECTED] wrote: On Sun, Mar 27, 2005 at 01:41:31PM +0100, Umar Sear wrote: Hi Guys, I am having trouble trying to get format_mp3 working to play music on hold. I have followed the instructions in the read-me and the wiki however it seems after un-installing mpg123, asterisk is not even attempting to play MOH. I don't do anything with music-on-hold, but... [moh_files] default = /var/lib/asterisk/moh-native ^^^ That spacing looks really odd... Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd Thanks, I wish it were that simple, adding removing the space makes no difference. Thanks Umar ___ 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] apps api?
On Mon, 28 Mar 2005 01:48:17 -0500, Jesse Guardiani [EMAIL PROTECTED] wrote: Hello, Is there a published apps API? Or do I need to just start reading source code? The wiki has some documentation but you need to look at the code and possibly network traces to better understand it. Umar ___ 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] codec for asterisk
G729 requires a licence from Digium. Umar On Thu, 24 Mar 2005 16:37:15 +0800, raymond [EMAIL PROTECTED] wrote: Hi, I had try to set up the call routing for asterisk to interwork with cisco AS5300 and found that Asterisk only support codec g711alaw and g711ulaw. For the other codecs (g723, g729, gsmfr), the calls were disconnected with cause value 63 (service option not available) or 127 (interworking error). Can anyone advise whether this is a restriction on asterisk? (or if I need to change anything on the standard config). Thanks. Raymond ___ 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] gsm cannot be found in any file form... but it's there
Are you sure it is in the right directory. Perhaps mentioning the directory you placed the file in and listing the section of your dialplan would go a long in helping someone help you. Umar On Fri, 18 Mar 2005 06:11:36 +, Scheda [EMAIL PROTECTED] wrote: Hey, I recorded this intro, and changed it to a gsm file in the shell, and I'm getting an error saying that it isn't in the directory at all when it's sitting right there. I don't know why that is. If you want to hear it, it's http://scheda.underfireradio.com/astintro.mp3 I don't know what the matter is, I've tried renaming it, copy and pasting it in there, deleting it and placing it back... I'm kinda out of ideas. ___ 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] newbie uk questions...
Checkout http://www.alwaysonvpn.com/ Umar. On Sun, 13 Mar 2005 11:21:02 +, Darrell Berry [EMAIL PROTECTED] wrote: hi: Just starting out with *, and I'm planning to heed the advice to start simple and small, but the goal i'm aiming for eventually is: *-based pbx for 10-20 seat small business, based in the UK. Users will have PoE SIP hardphones. So far so good, but two questions, both UK-specific, relating to connection to the outside world (PSTN or VoIP): - are there any UK-based VoIP providers targetting small business users: by which I mean support for multiple simultaneous connections in and out on the same DDI (to simulate traditional multi-channel ISDN PBX capabilities), and guaranteed SLAs/professional support? If so, has anyone dealt with any of them and do you have any recommendations (either for or against?). This includes ISPs getting into the VoIP arena. - failing that, what my options for *-compatible, UK-legal interconnections between a *-based PBX and UK PSTN? I'm looking for more channels than I will get from ISDN-2e, but less than ISDN-33 (probably): enough for say 4-8 simultaneous incoming/outgoing calls. I admit this is the area I'm least clear on! Even better: has anyone actually implemented either of these scenarios in the UK? Any feeeback/cheatsheets? Thanks - Darrell ___ 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] How can I eveluate trailing numbers in extensions.conf?
Checkout
http://www.voip-info.org/wiki-Asterisk+variables
I believe that should have the answer for you.
furthermore assuming that your number is always going to be 12 digits.
exten = _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number.
Hope this helps.
Umar
On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz [EMAIL PROTECTED] wrote:
Hi,
this message seems to not have made it to the list the first time - sorry
if it did.
My SIP provider includes trailing numbers to my account just fine, like
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc62e.644d2d75.0
From: Anonymous sip:[EMAIL PROTECTED];tag=as4b25d20f
Call-ID: [EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];tag=as406f4254
CSeq: 102 ACK
User-Agent: sipgate ser
Content-Length: 0
where 498645342456 is my SIP account phone number that can be reached from
the outside just fine. My question is, how can I evaluate the trailing 2
in my extensions.conf? This would be ideal for direct dialing to an attached
phone, and not be restricted to a single digit.
asterisk-dev The full number is included in the SIP message but does * keep
it
somewhere internally so that one could maybe add another externally usable
* variable? I browsed the source code but could not find anything... TIA!
Ciao,
hm
--
Today is the first day of the rest of the mess
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[Asterisk-Users] Asterisk Management API
Hi all, I am trying to write an application to monitor queues using the Asterisk Management API. So far I have had some level of sucess, basically reverse engineering the protocol and the event messages using ethereal etc. I know there are a couple of pages on the Wiki that attempt (no dis-respect to who ever did it as it has been a great help) to document the API and was wondering if there is more information available. Any pointers will be greatly appreciated. I hope to document my findings on the Wiki once I have definative information. Thanks Umar ___ 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 ${EXTEN} variable changes after Goto ?
I am pretty sure the answer is yes.
Umar
On Fri, 4 Mar 2005 00:33:46 +0100, Robert Rozman [EMAIL PROTECTED] wrote:
- Original Message -
From: Umar Sear [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 03, 2005 11:01 PM
Subject: Re: [Asterisk-Users] Why ${EXTEN} variable changes after Goto ?
On Thu, 3 Mar 2005 22:11:23 +0100, Robert Rozman [EMAIL PROTECTED]
wrote:
Hi,
I'm trying to implement dynamic routing of incoming calls to local
extension
if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand why ? How to
workaround on this one ?
Thanks in advance,
regards,
Rob.
[outbound-capi-ISDN]
exten = _0.,1,NoOp(Calling ISDN number ${EXTEN:1} on CAPI/7104370 from
${CALLERIDNUM})
exten = _0.,2,Dial,CAPI/7104370:b${EXTEN:1}|10|Tt
exten = _0.,3,Goto(s-${DIALSTATUS},1)
exten = _0.,103,NoOp(Calling ISDN number ${EXTEN:1} on CAPI/7104371)
exten = _0.,104,Dial,CAPI/7104371:b${EXTEN:1}|30|Tt
exten = _0.,105,Goto(s-${DIALSTATUS},1)
exten = _0.,205,Macro(outisbusy)
exten = s-NOANSWER,1,NoOp(NOANSWER - Setting dynamic autoroute for ISDN
number ${EXTEN:1} to local ext. ${CALLERIDNUM})
exten = s-NOANSWER,2,DBput(DYNAMIC/${EXTEN}=${CALLERIDNUM})
exten = s-NOANSWER,3,Congestion
exten = _s-.,1,Congestion
exten = _s-.,2,Macro(hangupcall)
Save the original extension to a variable like ...
exten = _0.,1,SetVar(myvar=${EXTEN})
Hi,
thanks for info. But another question arises - will this variable be unique
if for instance two or more calls happen in the same time ? I guess this is
more general Asterisk behaviour question...
Regards,
Rob.
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Re: [Asterisk-Users] Why ${EXTEN} variable changes after Goto ?
On Thu, 3 Mar 2005 22:11:23 +0100, Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
I'm trying to implement dynamic routing of incoming calls to local extension
if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand why ? How to
workaround on this one ?
Thanks in advance,
regards,
Rob.
[outbound-capi-ISDN]
exten = _0.,1,NoOp(Calling ISDN number ${EXTEN:1} on CAPI/7104370 from
${CALLERIDNUM})
exten = _0.,2,Dial,CAPI/7104370:b${EXTEN:1}|10|Tt
exten = _0.,3,Goto(s-${DIALSTATUS},1)
exten = _0.,103,NoOp(Calling ISDN number ${EXTEN:1} on CAPI/7104371)
exten = _0.,104,Dial,CAPI/7104371:b${EXTEN:1}|30|Tt
exten = _0.,105,Goto(s-${DIALSTATUS},1)
exten = _0.,205,Macro(outisbusy)
exten = s-NOANSWER,1,NoOp(NOANSWER - Setting dynamic autoroute for ISDN
number ${EXTEN:1} to local ext. ${CALLERIDNUM})
exten = s-NOANSWER,2,DBput(DYNAMIC/${EXTEN}=${CALLERIDNUM})
exten = s-NOANSWER,3,Congestion
exten = _s-.,1,Congestion
exten = _s-.,2,Macro(hangupcall)
Save the original extension to a variable like ...
exten = _0.,1,SetVar(myvar=${EXTEN})
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[Asterisk-Users] Zap timing device
Dear list, I have been using asterisk for some time now. However I have never used it with any of the digium or compatable cards (Purely used for SIP). I understand that for using Meetme, I need to have a timing device, which could either be hardware or zrdummy etc (I am not using any right now). Can someone tell me if the timing device is needed for voicemail and other applications too?. I am currently using voicemail, however at times the sound gets choppy and when I log on to the cli I see lots of event scheduled in the past warnings. I am using a fairly high speced machine (Athlon 64 3Ghz with 1GB RAM ...) so I can't imagine the system is being starved of CPU resource (Top shows the CPU idle rarely fallng below 90%) Currently I am not running stable versoin, rather a farely old cvs version (these are production units so hard to upgrade without a lot of effort), I do plan to upgrade to stable version as soon as possible, however would like some advice as to the best way to improve things and eliminate the choppyness and the warnings. Thanks Umar ___ 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] Zap timing device
On Tue, 22 Feb 2005 20:11:18 -, Kevin Walsh [EMAIL PROTECTED] wrote: Umar Sear [EMAIL PROTECTED] wrote: I have been using asterisk for some time now. However I have never used it with any of the digium or compatable cards (Purely used for SIP). I understand that for using Meetme, I need to have a timing device, which could either be hardware or zrdummy etc (I am not using any right now). There's also zaprtc to consider. See here: http://www.voip-info.org/wiki-Asterisk+timer Can someone tell me if the timing device is needed for voicemail and other applications too?. No - the timer is only required for conferences, IAX trunking and MOH. There might be others I've missed, but Voicemail is not one of them. Thanks, I really appreciate the feedback. Any ideas how I can the resolve the choppyness and scheduled in the past .. warnings. Umar -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] IVR option problem
Sounds like a DTMF issue, Check on the console when you dial-in externally, and make sure asterisk is reading the DTMF that you enter. Umar On Mon, 2004-10-18 at 15:19, ismaelg wrote: Hello all, I'm trying setting up an IVR on a Asterisk Soho PBX. My problem is when I dial the IVR extensión from an Asterisk internal extension all goes well, but when I dial the external number of the IVR, e.g. 119235656, the PSTN number of my asterisk, I get the same IVR menu but when I press on my phone the 1, to select the fist IVR option, or 2, to select the second one, (the IVR has only two options), I can't hear anything more on my phone. My phone gets silent. And I lost the rest of IVR locution. Regards. Ismael. Gil. Steven Critchfield wrote: On Mon, 2004-10-18 at 13:35 +0200, [EMAIL PROTECTED] wrote: Hi. Is possible to caprure calls with asterisk? I have a calling from onde device to another. While it´s ringing I´d wish to capture the calling from another device which has permissions to make it. is it possible? Check out pickup groups. BTW Digest users should be strongly urged to convert to normal messages as your less likely to make stupid mistakes with regards to responses. Like when you forget to TRIM irrelavent sections of the message, it doesn't force us all to rereceive large numbers of messages. __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Problems with voice menu
--- ismaelg [EMAIL PROTECTED] wrote: Hello all, I having a lot of troubles to configure a simple voice menu. In extensions.conf I have the following. [incoming] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,DigitTimeout,10 exten = s,4,ResponseTimeout,20 exten = s,5,Background(itranser/msg_bienvenida) exten = 1,1,Goto,contexto_extensiones exten = 2,1,Goto,contexto_operadora The context refered by the menu. (each context play me a diferent message only ) [contexto_operadora] exten = 2,2,Background(itranser/trans_operadora) exten = 2,3,Dial(SIP/ismael,s,1) [contexto_extensiones] exten = 1,1,Background(itranser/msg_pasar_ext) My problem, is when I touch the key 1 in my phone, after the msg_bienvenida, asterisk do not pass me to the correct context [contexto_extensiones]. Asterisk do not pass me to any context, asterisk do nothing when I press the 1 key on my phone. Have I missed something in my extensions.conf? or in sip.conf? I think this exten = 1,1,Goto,contexto_extensiones Should be exten = 1,1,Goto(contexto_extensiones,1,1) Umar Sear ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] asterisk make
On Tue, 2004-09-14 at 05:41, Dinesh wrote: cd ../asterisk # make clean; make install Hello when I do a make clean and make install, I get this error message on my asterisk box. bdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv -lssl /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Error 1 Any ideas? Dinesh. Am I missing something ? or are you forgetting to do a make before the make install ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Checking Return Codes
On Tue, 2004-09-07 at 21:48, Glenn A. Thompson wrote: Hi, I must be blind, how does one check then act upon the return code from the previous command? For instance, Answer says it can return non zero. How do I check for that. It doesn't set any other variables like Dial does. Most commands return a 0 or non zero value, and jump to priority n+101 if the return value is non-zero. In this case, all I'm really trying to do is not answer if the line has already been picked up. Do I have to make sure the channel is available before I issue the Answer cmd. Thanks, Glenn ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] X-Lite Meetme problem
On Wed, 2004-09-08 at 09:43, Vladyslav wrote: HI! Have a weird problem with X-lite Meetme. When X-Lite user are join to conference room NOT first one, than X-Lite user do not hear anything. This problem gone when X-Lite user get into conference room first (when nobody there). sip.conf [104] context=VoIP-only type=friend username=104 secret=test host=dynamic dtmfmode=rfc2833 mailbox=104 canreinvite=no disallow=all allow=ulaw ;allow=alaw ;allow=gsm On * console have such messages: when X-Lite using ULAW: Sep 8 09:47:17 WARNING[1233853360]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/4) on x-lite ALAW Sep 8 09:49:12 WARNING[1235344304]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/8) on x-lite GSM Sep 8 09:51:50 WARNING[1233804208]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 2 (read/write = 64/2) Please advice. Just a long shot, Try disallowing GSM for [104] Umar ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Caller id and the number of rings
On Wed, 2004-09-08 at 04:43, HengWee Chin wrote: Hi all, I have the following setup PSTN - ASTERISK - IVR (using dialogic card) 1) Caller id information is presented to asterisk during the first and second ring. 2) Hence, Asterisk waits for 2 rings before pickup the call and forwarding to the appropriate FXS port. 3) The IVR application also waits for 2 rings before picking up the call to get the caller id. 4) Hence any caller calling to the IVR will have to wait for 4 rings before he is serviced. This is too long. 5) Anyone have any idea how can I reduced the number of rings and still have caller id available to IVR? 6) If I were to switch PRI ISDN, would I still have the same problem? Thanks in advanced. Regards, Chin _ Hi Chin Did you find a resolution, I could not see a response that solves your issue. If you still have the issue it might be worth posting parts of your dialplan. Generally speaking, I would change the IVR part of your dialplan so that it checks if caller ID has been set, in which case it answers straightaway, if not then it waits for two rings. Umar ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Linux distribution
On Sat, 2004-09-04 at 22:53, Paul Mahler wrote: Asterisk should run well with any Linux distribution. Mepis, www.mepis.org, is pre-configured for * and might make your installation faster and easier. Paul Can you elaborate on what you mean by pre-configured. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Problem with HasNewVoicemail()
Try to specify the the context, it seems to be using default which may or may not be right. exten = s,1,HasNewVoicemail([EMAIL PROTECTED]|NEWMSGCOUNT) Umar On Thu, 2004-09-02 at 12:51, Nick Barnes wrote: Hi all, Maybe I'm being thick here, but I've had a look through the mailing list and the Wiki, and I can't seem to see details of anybody else with this problem With the following line: exten = s,1,HasNewVoicemail(201) I am getting the following error: -- Executing HasNewVoicemail(SIP/201-2f1e, 201) in new stack Sep 2 12:41:09 NOTICE[819221]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 201 at /var/spool/asterisk/voicemail/default/201/(null) does not exist Sep 2 12:41:09 WARNING[819221]: ast_expr.y:474 ast_yyerror: ast_yyerror(): syntax error: parse error; Input: 0 + ^ ^ And if I add the optional variable name to put the new count into: exten = s,1,HasNewVoicemail(201,NEWMSGCOUNT) The error message is an even more puzzling: -- Executing HasNewVoicemail(SIP/201-3277, [EMAIL PROTECTED]|NEWMSGCOUNT) in new stack Sep 2 12:45:33 NOTICE[851989]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 201 at /var/spool/asterisk/voicemail/default|NEWMSGCOUNT/201/(null) does not exist Sep 2 12:45:33 WARNING[851989]: ast_expr.y:474 ast_yyerror: ast_yyerror(): syntax error: parse error; Input: 0 + ^ ^ Which seems to be taking the variable name as part of the mailbox path. I have tried various combinations of ',' and '|', changing the mailbox to '[EMAIL PROTECTED]' and also surrounding parts with '', but the errors are all the same. The path '/var/spool/asterisk/voicemail/default/201/' definitely exists. The Asterisk version is - CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a Has anybody else seen this error or knows what stupid mistake/assumption I've made? Nick Barnes Senior IT Consultant. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Newbie - Voicemail Password Help
On Tue, 2004-08-31 at 10:53, Stephen Hon wrote:
Paul,
What you can do is modify the source code for the voicemail application.
Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the file
'vm-password' to 'pls-enter-vm-password'.
Recompile and install.
Then in your macro remove the line that plays the 'pls-enter-vm-password' file.
Steve
Why do that ? when you can simply replace the prompt file. Using your
method will need a recompile every time a different prompt needs to be
used.
From: [EMAIL PROTECTED] on behalf of Java Rockx
Sent: Mon 8/30/2004 8:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Newbie - Voicemail Password Help
Hello All.
I'm just beginning with Asterisk and I have it all working now. I'm using
Asterisk 1.0 RC1.
My only question is this; when I check my voice mail the PBX simply says
password. I wanted to make it say please enter your voice mail password so
I am using Background(pls-enter-vm-password).
However now I hear Please enter your voice mail password password when I
check my messages.
That's not a type-o. It says password twice.
Here is my extensions.conf file.
[macro-vmanswer]
exten = s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
exten = s,2,Background(pls-enter-vm-password)
exten = s,3,VoicemailMain(${ARG1})
exten = s,4,Hangup
exten = s,5,Voicemail(u${ARG1})
exten = s,6,Hangup
[default]
exten = 1002,1,Macro(vmanswer,1002)
The whole point of the vmanswer macro is to go to the voice mail main menu
automatically when calling from your own phone, otherwise it sends callers to
the voice mail system to leave a message. Perhaps there's a better way to do
this as well. If so, please let me know.
Regards,
Paul
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RE: [Asterisk-Users] G729 licenses
On Mon, 2004-08-30 at 16:26, Kevin Walsh wrote: Brian Wilkins [EMAIL PROTECTED] lazily top-posted: Point is that unfortunately many systems do use G 729 so it is necessary, in order to be compatible with existing gatekeepers, to use that codec. I'd love to use GSM but the existing systems do not support it. It is too costly to re-do everything, therefore you have to work around with what you got. Then you just dump the G.729-only supplier and find one that supports everything else. There are plenty to choose from. I am sorry, but you make it sound a lot simpler than it is. Bottom line is than in reality not everyone has the choice. Without even depending on any supplier(s), I would have issues using gsm, ilbc etc, etc. with our existing equipment. Simple solution dump the $100,000s equipment and find some that does support gsm etc etc. It does not work like that, people have to make commercial decisions and more often than not, that involves compromises. So in it's spirit I would agree with what you say, however in practise ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Asterisk Users] Help with SIP Hosted Billing Service
Not sure if I can help you on your dial-plan, however there are several
prepaid solutions for asterisk
I Developed my own using pascal and AGI as I requirements were quite
specific.
Umar
On Tue, 2004-08-24 at 23:20, E Samuels wrote:
Hello All,
I am trying to connect to a third partys SIP based Hosted Billing
service that supports G729 as I have found it difficult to get hold of
prepaid billing solution for Asterisk. I was able to get an IAX2
connection working fine with another provider but as I am unable to
find a billing service that supports IAX/IAX2, I need to connect to a
hosted billing service using SIP and G729.
I have purchased and installed the G729 licenses from Digium as
required.
My endpoints are Xten Pro and Snom.
I am not sure what the problem is but this is my dial plan in
extensions.conf
[default]
include = incoming
include = from-iax
include = from-sip
include = iaxtel-outbound
include = sip-pstn
include = iax2-pstn
[sip-pstn] **I am not certain if this is correct.
exten = _81NXXNXX,1,SetCallerID(123456)
exten = _81NXXNXX,2,SetCIDName(Myself)
exten = _81NXXNXX,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _81NXXNXX,4,Congestion
[iax2-pstn]
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
If anyone has any suggestions I would be happy to listen.
Thanks in Advance.
Errol
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Re: [Asterisk-Users] Re: newb question regarding DTMF
Check what DTMF mode you are using in sip.conf and confirm that it matches what you have configured on x-lite. Search the list archive, this issue has been thrashed out several times and is probably documented on voip-info.org. Umar. On Tue, 2004-08-24 at 17:03, Erik Anderson wrote: On Mon, 23 Aug 2004 18:20:58 -0500, Erik Anderson [EMAIL PROTECTED] wrote: Hello all - I'm just starting to play around w/ asterisk, and I've run into a seemingly simple problem that has really manged to frustrate me... I'm running the latest cvs version of *, and am trying to dial in to the default extention 1000 demo using x-lite. I can dial and hear the greeting no problem, but when I try and send any DTMF tones, I don't get any response. Is there something specific I need to set in my sip.conf to allow DTMF? Bump Any advice? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Voicemail Couldn't read username error
The error you are getting suggest that Asterisk can't read the DTMF being entered. Umar On Tue, 2004-08-24 at 18:49, Bill wrote: Hi, I have Asterisk running with the VoiceMail. Using the latest CVS. I have my extensions.conf setup so that if a SIP caller dials *99 the VoicemailMain() as follows: exten = *99,1,Wait(1) exten = *99,2,VoicemailMain() A couple days ago I installed the MySQL/Voicemail support described at http://www.voip-info.org/wiki-Asterisk+voicemail+database Now for some reason when I call *99 from a SIP extension I am prompted for a Mailbox number but then nothing happens. It used to prompt for a password. I've un-installed the MySQL/Voicemail support and it still doesn't work. From the Asterisk console I get the following message when I hangup when I am NOT prompted for a password. Aug 24 12:34:36 WARNING[319509]: app_voicemail.c:3568 vm_execmain: Couldn't read username Any ideas? I can see on several lists that this is a common problem but I haven't found the answer yet. Bill Dunn ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] External MW Lamp On/Off
I have done something simmillar, but not the same. I send mwi notification to our softswitch (SIP). Basically I wrote a small app in pascal that sends a sip message to the softswitch. The app is called everytime a message is left or retrieved, using the extrennotify option in voicemail.conf. You could easily do something simillar, what you need to do, is write a script or app (if one does not already exist) that creates call file based on the parameters passed by externnotify. Hope this helps. Umar --- Greg Blakely [EMAIL PROTECTED] wrote: One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp when the message has been played. I understand that this has something to do with a file that is placed in /var/spool/asterisk/outgoing, but I have no idea about + what the contents of that file should be, + how Comedian mail would initiate putting the file into the outgoing queue, and + how Comedian mail would initiate putting the 'extinguish' file into the outgoing queue. Has anyone done this sort of thing already? If so, can you point me in the right direction? As I mentioned in yesterday's post, I did find a question and partial answer to this in the asterisk-users archives, but I need a bit more information before I can make it work for me. Thanks in advance for any help you can give me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 voicemail from mysql no longer working
Hi All, I hope someone can help. I have a system that I have recently upgraded to latest CVS and my voicemail is not working from mysql database. I get an error on the console saying No entry in voicemail config file for 'number' whilst there is an entry in the database for the specified number. It seems like app_voicemail is no longer checking the database even though I can see that it is enabled and logs in when asterisk starts. I am sure I am missing something very basic, but could not find what ! Please help. ___ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk voicemail from mysql no longer working
Further invetigation revealed that app voicemail did not like the fact that I had the context set to 'local' as apposed to 'default' Any ideas' or shall I raise this as a bug ? Umar. --- Umar Sear [EMAIL PROTECTED] wrote: Hi All, I hope someone can help. I have a system that I have recently upgraded to latest CVS and my voicemail is not working from mysql database. I get an error on the console saying No entry in voicemail config file for 'number' whilst there is an entry in the database for the specified number. It seems like app_voicemail is no longer checking the database even though I can see that it is enabled and logs in when asterisk starts. I am sure I am missing something very basic, but could not find what ! Please help. ___ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] FINALLY! a good book about Asterisk.
--- Andy Powell [EMAIL PROTECTED] wrote: On 08/07/2004 at 22:19 usedcanon wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: 08 July 2004 20:15 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk. what does that have to do with an overpriced book? and i agree with Joe. With this book sourcing most of the documentation directly from wiki, why pay for something thats free? Id rather donate $49 to keeping wiki free to the enviroment. I second that, I think a more reasonably priced book in PDF fromat would have been better. Umar. Late last year I was approached by a publisher asking if I would be interested in writing an asterisk book. I said a polite no (after some discussion) for a number of reasons: 1. Precisely what the author of this book is experiencing. Being bitchslapped by the asterisk community, for no apparent good reason. Since very few of the people on this list have actually read the book this early critism and mud slinging appears unfounded. Let's face it - the biggest failing of asterisk is it's lack of documentation. Sure there are guides, documentation projects.. but all of these rely on people giving up their free time... and since we don't have much of that, progress is slow. Anything that helps document asterisk and how to get it set up can't be all that bad. 2. I hand't heard of the publisher before, and a google search didn't turn up the most favourable links. 3. Asterisk changes day by day.. If I'd gone with it the book would have been out by now and (aside from being bitchslapped) I'd probably immediately have had to start a 2nd edition.. I'm not a writer... I can't even spell properly. I don't know what the author was offered, but if it was just 15% then perhaps the deal I was offered wasn't as bad as I thought... At $49 it is quite expensive, however, when funds allow I'll more than likely buy a copy out of interest - I consider myself fairly a well seasoned asterisk person, but hey it might teach me something too... I'm prepared to give it a chance. All IMHO of course... Andy You make some very valid points. And to be honest I applaud the authors effort in documentation. However, the price etc I think is a bit on the high side. For something that with the best effort is not going to be complete, simply due to the fluid nature of asterisk, I think it would have been better to have had a PDF offering with a promise of updates for a certain period. Same as you, all of course IMHO ... On a separate note, I would like to see those criticising (myself included) to make an effort and write a book themselves. Umar. ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Minimum install required for Asterisk + voicemail SIP friends from mysql
I have been trying to install asterisk with MySQL for voicemail and SIP friends. Using redhat 9 I am installing all the base components required for asterisk, mysql, mysql server and mysql-devel. If I do a make clean, make install without enabling the mysql options in the /apps/Makefile and /channels/Makefile all goes well and make completes. If I enable the options Make fails, when it gets to -L/usr/lib/mysql -lmysqlclient -lz with an error /usr/bin/ld: cannot find -lz Now If I do the same after installing Redhat 9 doing a complete install make completes successfully. Ideally I would (I am sure others would too) like to install the minimum required. Help/advice will be greatly appreciated. Umar. ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Do people actually answer questions here?
Hi Andrew, I sympathise with your opinion. However if someone was to analyse the messaged in the list they would find that the most basic of questions get most replies. I mean those questions that would take a few minutes to answer searching through the wiki or google. Where questions that are not so straighr forwared get ignored. In my own experiance, every question that I have posted (after hours if not days of searching) has gone ignored. I must add, at no stage though have I felt a reason to complain, as even without answering any of my questions, this list has given me a wealth of knowledge. Thanks Umar. --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 29 June 2004 02:42, Ralf Van Dooren wrote: But instead of complaining about the lack of -findable- documentation, one can try to enhance existing documentation. That's the power of Open Source. As I am not a coder, I'll be trying to help the community by making the documentation better, especially for 'new bees' like me. This is where I have a problem. The documentation is centered on voip-info.org and on Asterisk's plainly marked Documentation link. It's been 32000 messages since I've signed up but I am *positive* that BOTH links are provided on the autoresponder when you sign up to this list. How crystal-effing-clear must things be for people to go and look for themselves before complaining on this list? Must we make the list moderated and autorespond with pre-fab Google searches that the asker simply has to click on and save themselves the trouble of writing the query into Google's search window themselves? I'm serious here -- voip-info.org's search engine works. Google works. For newbies yes they may have some trouble with the incantations but that's why they should not be diving in headfirst and then bitterly complaining that nothing works on the list. READ, dig around voip-info and asterisk's site. There's a WEALTH of knowlege there and most of it is not cryptic. A lot of it is even geared to newbies. Perhaps a glossary would be helpful but every day I start to think that basic research skills should be a prerequisite before being allowed to play with any OSS project. It's frustrating for the newbie, frustrating for the experts and all around a bad thing. You'll never have enough documentation to satisfy some people because they don't want to educate themselves; they want to ask questions and get personalized answers. We do that too, for a price. That's what consulting is all about. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] using SetCDRUserField in an AGI script
Hi I am trying to use SetCDRUserField in an agi script
but with no success.
I am using the CDR mysql addon, however I can't see it
being at fault as my attempt is not doing anything to
the CVS CD either.
has anyone used this, any hints guidence would be
greatly appreciated.
The syntax I am using is like so ..
res=DoExec('SetCDRUserField','12345');
and then dialing the relevant extention.
Thanks
Umar.
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Re: [Asterisk-Users] making * more like a normal pbx
How about cmd DISA ? Umar --- Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2004-06-14 at 04:54, Jacob Hunter wrote: once u press 9 is there a way to make it so it restores dial tone,like most pbx's do? so dial tone , 9, dialtone, then ur local num Look at ignorepat= -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please help !!!! - IAX, MYSQL - Cant make calls
Hi, I apologies for reposting this message, I am getting no where in solving this issue. And I am sure it is something very simple. I have two Firefly clients configured, If I use iax.conf to specifiy the accounts everything seems to work as expected. However If I use mysql, I can register and recieve calls on the firefly accounts (from SIP etc) but can not make calls between the two or anything else. I get a message on firefly Call ended with xxx reason : no authority found On the console I see the following message CHAN_IAX2 ... Socket_Read: Rejected connect attempt from IP Please help, Umar. Yahoo! Messenger - Communicate instantly...Ping your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Iax2 ringtone problem
Hi Jean, It seems that no one on the list is interested in IAX, I have posted a couple of basic questions but no ones seems to want to answer. I guess everyone is busy right now. Anyway back to your question. When you say the ringtone , do you mean the rinback tone (what the caller hears) or bell to notify the callee that there is a call. Umar. --- Jean-Francois Dubé [EMAIL PROTECTED] wrote: Hi, i have a problem with iax2 and ringtone. Here is the call path pstn - asterisk - iax - firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really appreciate on how i can fix my iax issue JF Thank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] FW: question about prepaid app_prepaid
Thanks to the lack of documentation, I decided to write my own AGI script (working but no where near complete) Look forward to replies and guidence on this topic. Umar. --- Yang Tao [EMAIL PROTECTED] wrote: Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk on Apple PPC with YDL
Very interesting, Would like to hear what sort of performance you get out of it. I was considering linux on a sun box. Anyone done that ? Umar. --- Darren Sessions [EMAIL PROTECTED] wrote: Fyi, Successfully compiled Asterisk on an Apple G4 PPC with Yellow Dog Linux - without any source modifications. Worked fast and smooth. - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Automating calls
Yes you can, I have never used it but here is a link http://www.voip-info.org/wiki-Asterisk+auto-dial+out Umar --- Simon [EMAIL PROTECTED] wrote: Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX, MYSQL - Rejected connect attempt from
Hi I am trying to use firefly as an IAX client with asterisk. If I populate the iax.conf with the user info, I can make calls successfuly. However if I use MYSQL and populate the records for each users I get an error saying Rejected connect attempt from 8.1.2.1 I am looked in the lists to see if there was something obvious that I am missing but could not find anything. Your help will be greatly appreciated. Thanks Umar. Yahoo! Messenger - Communicate instantly...Ping your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] AGI Pascal
Hi Andy, Once again thanks. This should make things a lot easier for me. I am greatful. btw what is the command line to execute the freepascal ide, also do you have any other recomendations. Thanks Umar. --- Andy Powell [EMAIL PROTECTED] wrote: On 28/05/2004 at 19:58 usedcanon wrote: Hi Andy, I am most certainly interested. If you have some example code using a DB (MySQL maybe) that would be extremelly helpful. BTW, I am new to fpc(Turbo pascal, Delphi and now Kylix), does it have a linux command line IDE like the DOS version Thanks for your help Umar Sorry umar, I missed your reply in the influx of messages...just spotted it...I'll tar it up and put it on my site. I've also got an example which connects to a mysql db (which I'll include) Yes there is an ide for linux... but e.. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger - Communicate instantly...Ping your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] AGI Pascal
hi Peter, Your feedback is greatly appreciated. Having not done any AGI before I was not sure what to expect. My requirements are very basic at the moment, and time as you say is money. my best option is to find something simmillar and customise it to my needs. Umar. --- Peter Corlett [EMAIL PROTECTED] wrote: usedcanon [EMAIL PROTECTED] wrote: Thanks, suddenly makes sense now. I guessed that is the case however was not sure. Any opinion on what is more/most efficient, using a scripting language like perl or a compile app in C/pascal. Define efficient. A C program would normally be expected to be about ten times faster than a Perl script. But when it's 10ms to execute instead of 100ms, it probably doesn't matter. If your time is not free, it may be more efficient to write a quick script in Perl and buy a faster server than it is to spend ages writing in C. Either way, if you're spending anything bit a trivial amount of CPU time executing AGI scripts (whatever the language), you've probably misdesigned something. So the ultimate answer is that AGI scripts should be written in whatever language you're most comfortable doing them in. -- Vice is its own reward. It is virtue which, if it is to be marketed with consumer appeal, must carry Green Shield stamps. - Quentin Crisp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger - Communicate instantly...Ping your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Time to lock down v1.1?
Hi Rich, Sounds like a good idea. Umar --- Rich Adamson [EMAIL PROTECTED] wrote: Isn't it about time to lock down added functionality to v1.1 and fix the remaining bugs? There has been a significant amount of traffic on the cvs list, the irc and other channels with folks spending time adding new functionality to Head. Think its time to lock it down, fix the bugs that have been introduced, and get to something that the _majority_ can agree to call v1.1 Stable in real production terms. It's a known fact that bugs are not being fixed in Stable, and even Mark has suggested no one should be running Stable in a production environment. There has been a number of postings in the last few days relative to bugs in sip, iax2, zaptel, codecs, etc. The add-on folks are obviously also having problems keeping up with modifying patches to a constantly moving target, and applying those to Stable is fruitless. I'd even suggest that no v1.2 Head be created until such time as the majority of bugs are fixed, and that souce _then_ copied to whatever the next version is going to be called. All in favor? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger - Communicate instantly...Ping your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Voicemail + SIP Message header
Title: RE : [Asterisk-Users] Voicemail + SIP Message header
Hi Deepak,
I had a similar setup, However I was able to configure the softswitch to send a prefix followed by the number dialled in the to field. This way I could route the call to the right mailbox and be able to play the right busy or away message.
Can you say what softswitch you are using ?
Umar
I am trying to use Asterisk as a pure voicemail system and have the following
setup:
I have the * setup as a SIP peer to a softswitch. When someone calls a number on the softswitch and no one picks up the phone, the softswitch forwards the call to the * using SIP. The message header of the SIP INVITE has the number originally called in the To: field, but the INVITE is still being sent to the number asterisk is configured for.
Is there any way that I can configure asterisk to read the To: field in the message header of the SIP INVITE and then go to the mailbox of the corresponding number?
Thanks
Deepak
Registered in England No. 04348334. Tel: (+44) 0118 965 5600
This message is subject to and does not create or vary any contractual relationship between alwaysON Group, its subsidiaries or affiliates ("Emperian alwaysON") and you. Internet communications are not secure and therefore alwaysON Group does not accept legal responsibility for the contents of this message. Any view or opinions expressed are those of the author. The message is intended for the addressee only and its contents and any attached files are strictly confidential. If you have received it in error, please telephone the number above. Thank you.
[Asterisk-Users] Asterisk with MySQL on Redhat 9
Hi I really hope somebody can help me out.
I have an asterisk installation working on a Redhat 9
system. I now want to add the MySQL functionally to it. However when I make the
necessary changes, (downloading the add-ons, and changing the Make file) the
make fails.
I have looked into this and I think I know what the problem
is. Basically I only have MySQL binaries installed. Can anyone advice me what
packages I need to install to get this going.
Help will be greatly appreciated.
Umar.
Registered in England No. 04348334. Tel: (+44) 0118 965 5600
This message is subject to and does not create or vary any contractual relationship between alwaysON Group, its subsidiaries or affiliates ("Emperian alwaysON") and you. Internet communications are not secure and therefore alwaysON Group does not accept legal responsibility for the contents of this message. Any view or opinions expressed are those of the author. The message is intended for the addressee only and its contents and any attached files are strictly confidential. If you have received it in error, please telephone the number above. Thank you.
[Asterisk-Users] Asterisk on FreeBSD
Hi there, Has anyone had much success installing Asterisk on FreeBSD 5 upwards? If so what are the packages required to get asterisk working. Thanks Umar. Registered in England No. 04348334. Tel: (+44) 0118 965 5600 This message is subject to and does not create or vary any contractual relationship between alwaysON Group, its subsidiaries or affiliates (Emperian alwaysON) and you. Internet communications are not secure and therefore alwaysON Group does not accept legal responsibility for the contents of this message. Any view or opinions expressed are those of the author. The message is intended for the addressee only and its contents and any attached files are strictly confidential. If you have received it in error, please telephone the number above. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 as voicemail only.
Hi,
I am trying to setup asterisk as a SIP based voicemail only
server. I am seeking advice regarding the best way to do this.
I have already got asterisk working as a voicemail server
and integrated with existing SIP based Centrex PBX. My next task is to
configure a live system and would like to strip out all the un-necessary stuff.
For instance I dont need support for any ISDN cards etc.
Also I would like some recommendation on OS platform to use,
again I am looking for a minimalist approach, not so much to save on disk
space, rather to minimize the active applications that are running.
Help and advice will be greatly appreciated.
Umar.
Registered in England No. 04348334. Tel: (+44) 0118 965 5600
This message is subject to and does not create or vary any contractual relationship between alwaysON Group, its subsidiaries or affiliates ("Emperian alwaysON") and you. Internet communications are not secure and therefore alwaysON Group does not accept legal responsibility for the contents of this message. Any view or opinions expressed are those of the author. The message is intended for the addressee only and its contents and any attached files are strictly confidential. If you have received it in error, please telephone the number above. Thank you.
[Asterisk-Users] Manually loading modules.
Hi Another question based on the same setup I mentioned before (* as a voicemail only server). By default * automatically loads several modules, I am trying to figure out the minimum list of modules that I need to load for asterisk to work as a SIP based voicemail only server I don't need any ISDN functionality. I can do without IAX, MGCP, etc My aim here is to streamline the installation so that only the necessary applications are loaded (sharing system resource) Thanks Umar. Registered in England No. 04348334. Tel: (+44) 0118 965 5600 This message is subject to and does not create or vary any contractual relationship between alwaysON Group, its subsidiaries or affiliates (Emperian alwaysON) and you. Internet communications are not secure and therefore alwaysON Group does not accept legal responsibility for the contents of this message. Any view or opinions expressed are those of the author. The message is intended for the addressee only and its contents and any attached files are strictly confidential. If you have received it in error, please telephone the number above. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
