[asterisk-users] ZAP and line disconnection detection
Dear Folks, Im looking for a way to detect if an analog line is connected to card or not (Im using Sangoma A200). Im using the dialtone detection when dialing but need a way to detect the disconnection of the line when it actually happens. Anyone have any hints or tricks for this? Regards. -- Mohammad Sh. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] limit concurrent calls on trunk supporting multiple DID
Thank you Alex, I'll handle this programatically if there is no other way. Best regards, Patrick On Thu, Sep 17, 2009 at 07:51, Alex Balashov abalas...@evaristesys.com wrote: You can set some kind of counter in the dial plan, call an AGI script, use func_odbc to make database calls, or otherwise achieve this programatically. -- Sent from mobile device On Sep 17, 2009, at 1:16 AM, Patrick asterisk-us...@ict-synergy.be wrote: Hello guys, I've one SIP trunk that support multiple DID. Only the trunk is documented in sip.conf (called DID is taken from the sip-header in real time). I would like to limit the number of simultaneous calls on each DID. Is there a way to achieve this ? My understanding is that the SIP configuration parameter limitonpeers will limit at the trunk level, right ? Thanks in advance Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
On Wed, 16 Sep 2009, Tilghman Lesher wrote: On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote: What g729 module should I download ? You should download only the licensed g.729 module from Digium, after paying a $10 license per concurrent user. All other modules have various problems (GPL violation or lack of paying the associated patent license). You cannot use G.729 without paying the license fee until after all associated patents expire (sometime in 2014). ... or be in in a country that doesn't honour software patents or copyrights - which the OP might well be... And there is a choice if the OP wants to pay the license fees - the OP does not have to dwonload the Digium one, there is another, competing one here: http://www.howlertech.com/products/howlets It's cheaper too. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ACR Anonymous Call Rejection
On Wed, 16 Sep 2009, Danny Nicholas wrote: What do you want your message to say? I'd just use busy-pls-hold and the caller would eventually get the idea that you weren't going to talk to them. You could also consider these Off-duty Not-auth-pstn Not-taking-your-call Number-not-answering tt-allbusy Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP and line disconnection detection
On Thu, Sep 17, 2009 at 09:34:56AM +0330, M Shokuie wrote: Dear Folks, Im looking for a way to detect if an analog line is connected to card or not (Im using Sangoma A200). Im using the dialtone detection when dialing but need a way to detect the disconnection of the line when it actually happens. I have no idea about the Sangoma drivers, but reecnt in-tree DAHDI drivers report this by raising a RED channel alarm if there's nothing connected. This means that Asterisk won't try dialing through it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR seleCtion
It's a bit off topic here (I would ask this on a QM or TB forum), but basically you redirect each IVR selection to a context where logging happens and then redirect to the queue. Just my two eurocents, l. 2009/9/16 Maria Cristina Bayno falls_m...@yahoo.com Hello Team, IVR selection of QUEUEMETRICS As we know queuemetrics had an IVR selection functionality where it can get the IVR keypress of a caller. We saw this link http://forum.queuemetrics.com/index.php?action=printpage;topic=503.0 and upon checking, its only determined the Queue, I want to get is the per IVR of a caller. Can you help me guys regarding this? I want to implement this with the trixbox asterisk. Any idea? Thank you Cristina Bayno Technical Support Bitstop Network Services, inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] custom voicemail e-mail
On Thu, 17 Sep 2009, Patrick wrote:
I was thinking also to replace the email sent by the voicemail by a php
script. My questions is simple, how do you manage to get the voicemail
variables from the php script ? Or, maybe, you get from stdin the
content of the email that should be send via sendmail ?
Unfortunately:
1) The command line is the value of the mailcmd variable from
voicemail.conf -- no additional arguments, nothing interesting like the
channel, exten, etc.
2) The environment is boring, nothing of value here.
3) Stdin is the header and body ready to be processed by sendmail -t or
your replacement. You could define emailbody to something like:
emailbody=VM_CALLERID=${VM_CALLERID}\nVM_CIDNAME=${VM_CIDNAME}\nVM_CIDNUM=${VM_CIDNUM}\nVM_DATE=${VM_DATE}\nVM_DUR=${VM_DUR\
}\nVM_MAILBOX=${VM_MAILBOX}\nVM_MSGNUM=${VM_MSGNUM}\nVM_NAME=${VM_NAME}\n
to make it somewhat easy to parse. If you need something else, you could
stuff it into CALLERID(name) before calling the voicemail application.
What do you want to accomplish?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] custom voicemail e-mail
Hello Steve,
Thats what I was expecting :-(
I want to send an email in html format as well as sending an SMS to
the mailbox owner using clickatell's api
Any other ways to do this ?
Best regards,
Patrick
On Thu, Sep 17, 2009 at 09:26, Steve Edwards asterisk@sedwards.com wrote:
On Thu, 17 Sep 2009, Patrick wrote:
I was thinking also to replace the email sent by the voicemail by a php
script. My questions is simple, how do you manage to get the voicemail
variables from the php script ? Or, maybe, you get from stdin the
content of the email that should be send via sendmail ?
Unfortunately:
1) The command line is the value of the mailcmd variable from
voicemail.conf -- no additional arguments, nothing interesting like the
channel, exten, etc.
2) The environment is boring, nothing of value here.
3) Stdin is the header and body ready to be processed by sendmail -t or
your replacement. You could define emailbody to something like:
emailbody=VM_CALLERID=${VM_CALLERID}\nVM_CIDNAME=${VM_CIDNAME}\nVM_CIDNUM=${VM_CIDNUM}\nVM_DATE=${VM_DATE}\nVM_DUR=${VM_DUR\
}\nVM_MAILBOX=${VM_MAILBOX}\nVM_MSGNUM=${VM_MSGNUM}\nVM_NAME=${VM_NAME}\n
to make it somewhat easy to parse. If you need something else, you could
stuff it into CALLERID(name) before calling the voicemail application.
What do you want to accomplish?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] How to configure a coverage pathfor anextension???
On Wed, 16 Sep 2009, Steve Edwards wrote:
On Wed, 16 Sep 2009, Danny Nicholas wrote:
I'd try this:
- exten = 4000,1,Dial(SIP/4000,20,ikKtT)
- exten = s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
- exten = s-NOANSWER,2,Voicemail(4000)
- exten = s-BUSY,1,Dial(SIP/4001,20,iKkTt)
- exten = s-BUSY,2,Voicemail(4000)
- exten = h,1,hangup
Don't you need a goto(s-${DIALSTATUS},1) in there somewhere?
BTW, everybody seems to do s-${DIALSTATUS}. Why not just
${DIALSTATUS}?
s- doesn't seem to add any value to me.
I suspect everyone is copying one example that was presented some years
ago ;-)
From the more than one way to skin a cat department, I do it this way:
exten = s,n,GotoIf($[(${DIALSTATUS} = CHANUNAVAIL) | (${DIALSTATUS} =
CONGESTION)]?:${DIALSTATUS})
...
exten = s,n(BUSY),Noop(We got BUSY)
exten = s,n,Busy()
exten = s,n(NOANSWER),Noop(We got NOANSWER)
exten = s,n,Congestion()
etc.
So Goto'ing a priority rather than an extension.
For the original poster, the simplest/crudest way is simply:
exten = s,n,Dial(SIP/4000,,10)
exten = s,n,Dial(SIP/4001,,10)
exten = s,n,Dial(SIP/4002,,10)
exten = s,n,Dial(SIP/4003,,10)
exten = s,n,Voicemail...
knowing that if one of the Dial's succeedes then the rest of them will not
action and if it fails to bridge for any reason execution will just carry
on to the next step...
Gordon
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Re: [asterisk-users] call-limit on dahdi channel
Alex Samad a écrit :
Hi
how do i set the call-limit on a dahi line - its connected to the pstn
network - shared fax line. How do i tell asterisk not to send more than
1 call there !
exten = _XXX.,20(Start),Set(GROUP()=PSTN)
exten = _XXX.,n,GotoIf($[${GROUP_COUNT(PSTN)}=0]?lineOpen)
exten = _XXX.,n,Congestion()
exten = _XXX.,n,Hangup(34)
exten = _XXX.,n(lineOpen),NoOp(Place your call to DAHDI channel)
--
Daniel
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Re: [asterisk-users] call-limit on dahdi channel
On Thu, Sep 17, 2009 at 08:18:13AM +1000, Alex Samad wrote: Hi how do i set the call-limit on a dahi line - its connected to the pstn network - shared fax line. How do i tell asterisk not to send more than 1 call there ! Asterisk will not send out more than one call on that line. You want to avoid calling if someone else is calling on a fax machine connected to the same line? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connected Line ID for Asterisk 1.4
Jeff LaCoursiere wrote: The last patch for RPID is marked for 1.4.23.1 (2/10/09) : https://issues.asterisk.org/file_download.php?file_id=21601type=bug I've been running it on 1.4.23.1 since Feb. Thanks! I'll give it a shot, Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR seleCtion
Hello team, Thanks Lenz, we actually did that. The ivr data capture at our end is working. We only want to capture one row per call. Is there any idea regarding this?? tHank you so much. Regards, Cristina --- On Thu, 9/17/09, Lenz Emilitri lenz.lo...@gmail.com wrote: From: Lenz Emilitri lenz.lo...@gmail.com Subject: Re: [asterisk-users] IVR seleCtion To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, September 17, 2009, 7:07 AM It's a bit off topic here (I would ask this on a QM or TB forum), but basically you redirect each IVR selection to a context where logging happens and then redirect to the queue. Just my two eurocents, l. 2009/9/16 Maria Cristina Bayno falls_m...@yahoo.com Hello Team, IVR selection of QUEUEMETRICS As we know queuemetrics had an IVR selection functionality where it can get the IVR keypress of a caller. We saw this link http://forum.queuemetrics.com/index.php?action=printpage;topic=503.0 and upon checking, its only determined the Queue, I want to get is the per IVR of a caller. Can you help me guys regarding this? I want to implement this with the trixbox asterisk. Any idea? Thank you Cristina Bayno Technical Support Bitstop Network Services, inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com -Inline Attachment Follows- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
On 09/17/2009 02:52 PM, Gordon Henderson wrote: On Wed, 16 Sep 2009, Tilghman Lesher wrote: On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote: What g729 module should I download ? You should download only the licensed g.729 module from Digium, after paying a $10 license per concurrent user. All other modules have various problems (GPL violation or lack of paying the associated patent license). You cannot use G.729 without paying the license fee until after all associated patents expire (sometime in 2014). ... or be in in a country that doesn't honour software patents or copyrights - which the OP might well be... And there is a choice if the OP wants to pay the license fees - the OP does not have to dwonload the Digium one, there is another, competing one here: http://www.howlertech.com/products/howlets It's cheaper too. Gordon Where do you think he might live? Antarctica? :-) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
On Thu, 17 Sep 2009, Steve Underwood wrote: Where do you think he might live? Antarctica? :-) Beirut, Lybia. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
Actually what you mean by cheaper? 10 cents? === Live rates at 2009.09.17 12:49:51 UTC 5.99 GBP = 9.89128 USD United Kingdom Pounds United States Dollars 1 GBP = 1.65130 USD 1 USD = 0.605584 GBP === -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Thursday, September 17, 2009 2:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 On Wed, 16 Sep 2009, Tilghman Lesher wrote: On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote: What g729 module should I download ? You should download only the licensed g.729 module from Digium, after paying a $10 license per concurrent user. All other modules have various problems (GPL violation or lack of paying the associated patent license). You cannot use G.729 without paying the license fee until after all associated patents expire (sometime in 2014). ... or be in in a country that doesn't honour software patents or copyrights - which the OP might well be... And there is a choice if the OP wants to pay the license fees - the OP does not have to dwonload the Digium one, there is another, competing one here: http://www.howlertech.com/products/howlets It's cheaper too. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing or Adding a Line to the Extensions.conf in Asterisk
I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version
1.400 and I am simply trying to configure into the Extensions.conf
script an entry that will add to the Auto-Attendant a line that will
allow a Caller to enter a 0 (Zero) will then ring the extension(s)
of the Operator to speak directly with the OPERATOR rather than
entering an Employees Extension.
Below is a partial copy of the Extensions.conf file:
[globals]
[incoming]
exten = _X.,1,GotoIfTime(8:00-11:59|mon-fri|*|*?opened,s,1)
; closed for lunch
exten = _X.,n,GotoIfTime(12:00-13:00|mon-fri|*|*?closed,s,1)
exten = _X.,n,GotoIfTime(13:01-17:00|mon-fri|*|*?opened,s,1)
exten = _X.,n,GotoIfTime(17:01-23:59|fri|*|*?closed,s,1)
exten = _X.,n,GotoIfTime(*|sat-sun|*|*?closed,s,1)
exten = _X.,n,Goto(closed,s,1)
[opened]
include=parkedcalls
exten =
s,1,MixMonitor(incoming-from_${CALLERIDNUM}-${TIMESTAMP}-${UNIQUEID}.wav
)
; Removed exten 201 and replaced with 238 on 8/24/06
;exten = s,n,Dial(Sip/201Sip/209Sip/211|20)
exten = s,n,Dial(Sip/238Sip/209Sip/211|20)
exten = s,n,Dial(Sip/227Sip/225Sip/213|20)
exten = s,n,Goto(ivr,s,1)
exten = _2XX,1,Macro(extensions,${EXTEN})
exten = 1234,1,Dial(Sip/phone1,20) ;Aastra 480iCT
[closed]
exten = s,1,Goto(ivr,s,1)
[ivr]
exten = s,1,Answer
exten = s,n,Set(LOOPCOUNT=0)
exten = s,n(begin),Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(aa_1)
exten = s,n,WaitExten(10)
exten = s,n,Goto(loop,1)
exten = #,1,Directory(default|internal)
exten = i,1,Playback(invalid)
exten = i,n,Goto(loop,1)
exten = t,1,Goto(loop,1)
exten = loop,1,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])
exten = loop,n,GotoIf($[${LOOPCOUNT} 2]?hang,1)
exten = loop,n,Goto(ivr,s,begin)
exten = hang,1,Playback(vm-goodbye)
exten = hang,n,Hangup
exten = _2XX,1,Macro(extensions,${EXTEN})
exten = 999,1,MusicOnhold
[internal]
include=parkedcalls
include=outgoing
exten = _2XX,1,Macro(extensions,${EXTEN})
exten = *97,1,VoicemailMain(${calleridn...@default)
exten = *98,1,VoicemailMain
exten = _*XXX,1,VoicemailMain(${calleridn...@default)
exten = 300,1,MeetMe(100)
exten = 678,1,Goto(ivr,s,1)
[macro-extensions]
exten =
s,1,MixMonitor(exten_${ARG1}-from_${CALLERIDNUM}-${TIMESTAMP}-${UNIQUEID
}.wav)
exten = s,2,Dial(Sip/${ARG1}|20)
exten = s,3,Voicemail(${ARG1}|u)
exten = s,103,Voicemail(${ARG1}|b)
Thanks,
Paul Torres
IT Systems Manager
The Children's Home of Lubbock
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Re: [asterisk-users] G729
On Thu, Sep 17, 2009 at 01:27:19PM +0100, Gordon Henderson wrote: On Thu, 17 Sep 2009, Steve Underwood wrote: Where do you think he might live? Antarctica? :-) Beirut, Lybia. Beirut in Lybia? I thought Lybia only managed to snach Tripoly from Lebanon. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing or Adding a Line to the Extensions.confin Asterisk
This is simple; in the [ivr] context, add
- exten = 0,1,Dial(SIP/238,20,KkTt)
A better way to handle this is to write a key in your asterisk database
and set up a context to allow changing the value
[ivr]
- exten = 0,1,Set(OPEREXT=${DB(Oper/ext)})
- exten = 0,2,Dial(SIP/${OPEREXT},20,KkTt)
Else where
exten = 199,1(start7),Playback(record/nightopext)
exten = 199,n,NoOp(executando - ${extensao} - )
exten = 199,n,BackGround(beep)
exten = 199,n,Read(digito,,3)
exten = 199,n,Gotoif($[ ${LEN(${digito})} != 3]?start7)
exten = 199,n,SayDigits(${digito})
exten = 199,n,Set(DB(Oper/ext)=${digito})
exten = 199,n,BackGround(vm-goodbye)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Torres
Sent: Thursday, September 17, 2009 8:13 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Changing or Adding a Line to the Extensions.confin
Asterisk
I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version
1.400 and I am simply trying to configure into the Extensions.conf
script an entry that will add to the Auto-Attendant a line that will
allow a Caller to enter a 0 (Zero) will then ring the extension(s)
of the Operator to speak directly with the OPERATOR rather than
entering an Employees Extension.
Below is a partial copy of the Extensions.conf file:
[globals]
[incoming]
exten = _X.,1,GotoIfTime(8:00-11:59|mon-fri|*|*?opened,s,1)
; closed for lunch
exten = _X.,n,GotoIfTime(12:00-13:00|mon-fri|*|*?closed,s,1)
exten = _X.,n,GotoIfTime(13:01-17:00|mon-fri|*|*?opened,s,1)
exten = _X.,n,GotoIfTime(17:01-23:59|fri|*|*?closed,s,1)
exten = _X.,n,GotoIfTime(*|sat-sun|*|*?closed,s,1)
exten = _X.,n,Goto(closed,s,1)
[opened]
include=parkedcalls
exten =
s,1,MixMonitor(incoming-from_${CALLERIDNUM}-${TIMESTAMP}-${UNIQUEID}.wav
)
; Removed exten 201 and replaced with 238 on 8/24/06
;exten = s,n,Dial(Sip/201Sip/209Sip/211|20)
exten = s,n,Dial(Sip/238Sip/209Sip/211|20)
exten = s,n,Dial(Sip/227Sip/225Sip/213|20)
exten = s,n,Goto(ivr,s,1)
exten = _2XX,1,Macro(extensions,${EXTEN})
exten = 1234,1,Dial(Sip/phone1,20) ;Aastra 480iCT
[closed]
exten = s,1,Goto(ivr,s,1)
[ivr]
exten = s,1,Answer
exten = s,n,Set(LOOPCOUNT=0)
exten = s,n(begin),Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(aa_1)
exten = s,n,WaitExten(10)
exten = s,n,Goto(loop,1)
exten = #,1,Directory(default|internal)
exten = i,1,Playback(invalid)
exten = i,n,Goto(loop,1)
exten = t,1,Goto(loop,1)
exten = loop,1,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])
exten = loop,n,GotoIf($[${LOOPCOUNT} 2]?hang,1)
exten = loop,n,Goto(ivr,s,begin)
exten = hang,1,Playback(vm-goodbye)
exten = hang,n,Hangup
exten = _2XX,1,Macro(extensions,${EXTEN})
exten = 999,1,MusicOnhold
[internal]
include=parkedcalls
include=outgoing
exten = _2XX,1,Macro(extensions,${EXTEN})
exten = *97,1,VoicemailMain(${calleridn...@default)
exten = *98,1,VoicemailMain
exten = _*XXX,1,VoicemailMain(${calleridn...@default)
exten = 300,1,MeetMe(100)
exten = 678,1,Goto(ivr,s,1)
[macro-extensions]
exten =
s,1,MixMonitor(exten_${ARG1}-from_${CALLERIDNUM}-${TIMESTAMP}-${UNIQUEID
}.wav)
exten = s,2,Dial(Sip/${ARG1}|20)
exten = s,3,Voicemail(${ARG1}|u)
exten = s,103,Voicemail(${ARG1}|b)
Thanks,
Paul Torres
IT Systems Manager
The Children's Home of Lubbock
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Re: [asterisk-users] G729
On Thursday 17 September 2009 01:52:26 Gordon Henderson wrote: On Wed, 16 Sep 2009, Tilghman Lesher wrote: On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote: What g729 module should I download ? You should download only the licensed g.729 module from Digium, after paying a $10 license per concurrent user. All other modules have various problems (GPL violation or lack of paying the associated patent license). You cannot use G.729 without paying the license fee until after all associated patents expire (sometime in 2014). And there is a choice if the OP wants to pay the license fees - the OP does not have to dwonload the Digium one, there is another, competing one That one violates the GPL. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice Playback cutting first word or so of audio file
When I call inbound with a cell phone (via SIP PSTN trunk) some of my prompts the first word is cut off. I'm assuming the prompt is needing to be transcoded on the fly and it's not getting transcoded fast enough. I did a file convert to create gsm versions (currently they are referenced in my dial plan with no extension Seem to have same problem. How do I determine which file it's playing, for one and what is the likely cause of my issue? Note: If I call from a landline, this problem doesn't exist. I'm assuming there is some way to determine if resources are being expended to transcode. ox: WAV Chunk fmt sox: WAV Chunk data sox: Reading Wave file: Microsoft PCM format, 1 channel, 8000 samp/sec sox: 16000 byte/sec, 2 block align, 16 bits/samp, 146966 data bytes sox: 73483 Samps/chans sox: Input file mysoundfile.wav: using sample rate 8000 size shorts, encoding signed (2's complement), 1 channel Samples read: 73482 Length (seconds): 9.185250 Scaled by: 2147483647.0 Maximum amplitude: 0.980347 Minimum amplitude:-0.980347 Midline amplitude: 0.00 Meannorm: 0.190652 Meanamplitude:-0.004612 RMS amplitude: 0.294026 Maximum delta: 1.554443 Minimum delta: 0.00 Meandelta: 0.096981 RMS delta: 0.163688 Rough frequency: 708 Volume adjustment:1.020 sox: Detected file format type: gsm sox: Input file mysoundfile.gsm: using sample rate 8000 size bytes, encoding gsm, 1 channel Samples read: 73440 Length (seconds): 9.18 Scaled by: 2147483647.0 Maximum amplitude: 0.999756 Minimum amplitude:-1.00 Midline amplitude:-0.000122 Meannorm: 0.179742 Meanamplitude: 0.000149 RMS amplitude: 0.277143 Maximum delta: 1.305664 Minimum delta: 0.00 Meandelta: 0.084535 RMS delta: 0.140480 Rough frequency: 645 Volume adjustment:1.000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] custom voicemail e-mail
On Thursday 17 September 2009 02:52:42 Patrick wrote: I want to send an email in html format as well as sending an SMS to the mailbox owner using clickatell's api Any other ways to do this ? Use the externnotify option in voicemail.conf. The arguments to this program are the context, extension, new-voicemails, old-voicemails, and (on newer versions) urgent-voicemails. Take the new-voicemails (a number), subtract 1, load the corresponding .txt file, and you have all the information available to a voicemail message, so you can send whatever emails you need to from here. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
On Thu, 17 Sep 2009, Hristo Benev wrote: Actually what you mean by cheaper? 10 cents? As I live in the UK, it's more. By the time I add on my greedy banks conversion fees, etc. then if I'm honest and add on any import duties/customs, etc. it's way more than just 10 cents. And I know you'd rather pay for something in canadian dollars as opposed to sterling, well it works the other way too. I'd rather pay in sterling than any kind of dollars. Gordon === Live rates at 2009.09.17 12:49:51 UTC 5.99 GBP = 9.89128 USD United Kingdom Pounds United States Dollars 1 GBP = 1.65130 USD 1 USD = 0.605584 GBP === -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Thursday, September 17, 2009 2:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 On Wed, 16 Sep 2009, Tilghman Lesher wrote: On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote: What g729 module should I download ? You should download only the licensed g.729 module from Digium, after paying a $10 license per concurrent user. All other modules have various problems (GPL violation or lack of paying the associated patent license). You cannot use G.729 without paying the license fee until after all associated patents expire (sometime in 2014). ... or be in in a country that doesn't honour software patents or copyrights - which the OP might well be... And there is a choice if the OP wants to pay the license fees - the OP does not have to dwonload the Digium one, there is another, competing one here: http://www.howlertech.com/products/howlets It's cheaper too. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
On Thu, 17 Sep 2009, Tzafrir Cohen wrote: On Thu, Sep 17, 2009 at 01:27:19PM +0100, Gordon Henderson wrote: On Thu, 17 Sep 2009, Steve Underwood wrote: Where do you think he might live? Antarctica? :-) Beirut, Lybia. Beirut in Lybia? I thought Lybia only managed to snach Tripoly from Lebanon. Ah yes. Was listening to something about Lybia at the time I was typing that. Oops. Lebanon, of-course! (Both start with the same letter!) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice Playback cutting first word or so of audio file
James Hankins wrote: When I call inbound with a cell phone (via SIP PSTN trunk) some of my prompts the first word is cut off. I'm assuming the prompt is needing You need to add a Wait(1) after the Answer() Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice Playback cutting first word or so of audio file
Just tried, perfect! Thanks. Jim On Sep 17, 2009, at 9:56 AM, Doug Lytle wrote: James Hankins wrote: When I call inbound with a cell phone (via SIP PSTN trunk) some of my prompts the first word is cut off. I'm assuming the prompt is needing You need to add a Wait(1) after the Answer() Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice Playback cutting first word or so of audio file
James Hankins wrote: When I call inbound with a cell phone (via SIP PSTN trunk) some of my prompts the first word is cut off. I'm assuming the prompt is needing On Thu, 17 Sep 2009, Doug Lytle wrote: You need to add a Wait(1) after the Answer() Or answer(1000). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP HEADER FROM: without CALLERID(name):: PART DEUX
Hi all, Several years ago there was a thread on this list about the behavior of Asterisk when there was an empty display-name field in the SIP FROM header: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg142835.html It seems as if the thread was not answered, so allow me to scratch the scab off one more time :) Is it possible to configure Asterisk to not modify the display-name field of the FROM header? In other words, if the field is empty on the received INVITE, keep it empty on the sent INVITE. If the field is not empty on the received INVITE, simply retain the contents of the field in the sent INVITE. We're looking to implement this behavior for all calls, private and non-private. Thanks, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] custom voicemail e-mail
On Thursday 17 September 2009 02:52:42 Patrick wrote: I want to send an email in html format as well as sending an SMS to the mailbox owner using clickatell's api Any other ways to do this ? On Thu, 17 Sep 2009, Tilghman Lesher wrote: Use the externnotify option in voicemail.conf. The arguments to this program are the context, extension, new-voicemails, old-voicemails, and (on newer versions) urgent-voicemails. Take the new-voicemails (a number), subtract 1, load the corresponding .txt file, and you have all the information available to a voicemail message, so you can send whatever emails you need to from here. voicemail.conf says If you need to have an external program, i.e. /usr/bin/myapp called when a voicemail is left, delivered, or your voicemailbox is checked, uncomment this. Can myapp differentiate between left, delivered, or checked? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Freepbx database
Hellos I am using freepbx and asterisk. I am writing an AGI script to edit the values in findmefollow table. The script will enable users to delete and add follow me numbers from their phones. I want it to enable users enable/disable follow me. I can't seem to find a value in the database that deals with enabling/disabling followme. Please help -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] custom voicemail e-mail
On Thursday 17 September 2009 09:13:20 Steve Edwards wrote: On Thursday 17 September 2009 02:52:42 Patrick wrote: I want to send an email in html format as well as sending an SMS to the mailbox owner using clickatell's api Any other ways to do this ? On Thu, 17 Sep 2009, Tilghman Lesher wrote: Use the externnotify option in voicemail.conf. The arguments to this program are the context, extension, new-voicemails, old-voicemails, and (on newer versions) urgent-voicemails. Take the new-voicemails (a number), subtract 1, load the corresponding .txt file, and you have all the information available to a voicemail message, so you can send whatever emails you need to from here. voicemail.conf says If you need to have an external program, i.e. /usr/bin/myapp called when a voicemail is left, delivered, or your voicemailbox is checked, uncomment this. Can myapp differentiate between left, delivered, or checked? No, not as such. However, if you listen to all your messages when you check your mailbox, and you don't leave any in the new folder, then when externnotify runs, there will be a 0 in the new-messages argument. Presumably, the user doesn't care about the difference between left and delivered (the second being what happens when a voicemail is forwarded from another mailbox). -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx database
Well, why not disable it from the GUI and see what changes -- this is sort of the wrong list, but maybe someone knows more fully. James Mutuku listmut...@gmail.com wrote: Hellos I am using freepbx and asterisk. I am writing an AGI script to edit the values in findmefollow table. The script will enable users to delete and add follow me numbers from their phones. I want it to enable users enable/disable follow me. I can't seem to find a value in the database that deals with enabling/disabling followme. Please help -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales Alternatives: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
On Thu, 17 Sep 2009, Tilghman Lesher wrote: On Thursday 17 September 2009 01:52:26 Gordon Henderson wrote: On Wed, 16 Sep 2009, Tilghman Lesher wrote: On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote: What g729 module should I download ? You should download only the licensed g.729 module from Digium, after paying a $10 license per concurrent user. All other modules have various problems (GPL violation or lack of paying the associated patent license). You cannot use G.729 without paying the license fee until after all associated patents expire (sometime in 2014). And there is a choice if the OP wants to pay the license fees - the OP does not have to dwonload the Digium one, there is another, competing one That one violates the GPL. The free one or the Howlets one? However I can't see how the binary blobs of patented code which digium sells doesn't voilate the GPL either. It's nice to have competition. Keeps you on your toes. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice Playback cutting first word or so of audio file
Steve Edwards wrote: Or answer(1000). I'd prefer this route! Thanks. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
On Thu, Sep 17, 2009 at 10:50 AM, Gordon Henderson gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote: On Thu, 17 Sep 2009, Tilghman Lesher wrote: The free one or the Howlets one? However I can't see how the binary blobs of patented code which digium sells doesn't voilate the GPL either. It's nice to have competition. Keeps you on your toes. Gordon Because Digium OWNS the Asterisk code, and they make an exception for their binary code, is their right as owners (copyright holders) of the code. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice Playback cutting first word or so of audio file
Steve Edwards escribió: James Hankins wrote: When I call inbound with a cell phone (via SIP PSTN trunk) some of my prompts the first word is cut off. I'm assuming the prompt is needing On Thu, 17 Sep 2009, Doug Lytle wrote: You need to add a Wait(1) after the Answer() Or answer(1000). Cool, didn't know about that one. One less line of code in the dialplan. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice Playback cutting first word or so of audio file
Yes, Steve could write a book. I would probably buy The Luddites guide to Asterisk. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: Thursday, September 17, 2009 10:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voice Playback cutting first word or so of audio file Steve Edwards escribió: James Hankins wrote: When I call inbound with a cell phone (via SIP PSTN trunk) some of my prompts the first word is cut off. I'm assuming the prompt is needing On Thu, 17 Sep 2009, Doug Lytle wrote: You need to add a Wait(1) after the Answer() Or answer(1000). Cool, didn't know about that one. One less line of code in the dialplan. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I'm not getting the ability to leave a voicemail-message
I'm having a little problem with voicemail. Actually I'm not getting the
ability to leave a voicemail-message.
This is part of the dialplan :
exten = s,n(voicemail),PlayBack(/var/lib/asterisk/sounds/voicemail/${ARG1})
exten = s,n,NoOp(${ar...@boxes)
exten = s,n,Voicemail(${ar...@boxes)
exten = s,n,Hangup()
exten = s,n,MacroExit
This is the CLI-output :
[Sep 17 17:23:06] -- Executing [...@macro-openingsuren:49]
NoOp(IAX2/zoiper-16222, 908...@boxes) in new stack
[Sep 17 17:23:06] -- Executing [...@macro-openingsuren:50]
VoiceMail(IAX2/zoiper-16222, 908...@boxes) in new stack
[Sep 17 17:23:06] -- IAX2/zoiper-16222 Playing 'vm-intro' (language 'be')
[Sep 17 17:23:06] == Spawn extension (macro-openingsuren, s, 50) exited
non-zero on 'IAX2/zoiper-16222' in macro 'openingsuren'
[Sep 17 17:23:06] == Spawn extension (092779077, s, 6) exited non-zero on
'IAX2/zoiper-16222'
[Sep 17 17:23:06] -- Hungup 'IAX2/zoiper-16222'
vm-intro is an empty file. I deleted the original and replaced it with a
touch vm-intro.gsm.
So Asterisk quickly goes on after 'playing' the sound-file and
immediately hangs up (which is the next priority to execute).
I don't understand why I am not connected with the voicemailbox to leave
a message ???
Greetingz,
Jonas.
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Re: [asterisk-users] I'm not getting the ability to leave a voicemail-message
On Thu, 2009-09-17 at 17:31 +0200, jonas kellens wrote: vm-intro is an empty file. I deleted the original and replaced it with a touch vm-intro.gsm. I'm curious as to why you did this. Why didn't you simply pass the 's' option to the VoiceMail() application to have it skip the introductory message? -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice Playback cutting first word or so of audio file
At 06:49 AM 9/17/2009, you wrote: When I call inbound with a cell phone (via SIP PSTN trunk) some of my prompts the first word is cut off. I'm assuming the prompt is needing to be transcoded on the fly and it's not getting transcoded fast enough. I did a file convert to create gsm versions (currently they are referenced in my dial plan with no extension I've always added a wait(1) as the first line in those plans with the problem and it goes away. I think it's that the prompt starts before the call is finished being set up. Actually, it might fix it but I did that for a completely different reason having to do with caller ID on a PSTN line. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
On Thursday 17 September 2009 10:01:46 Moises Silva wrote: On Thu, Sep 17, 2009 at 10:50 AM, Gordon Henderson gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote: On Thu, 17 Sep 2009, Tilghman Lesher wrote: The free one or the Howlets one? However I can't see how the binary blobs of patented code which digium sells doesn't voilate the GPL either. Because Digium OWNS the Asterisk code, and they make an exception for their binary code, is their right as owners (copyright holders) of the code. Close enough. Digium doesn't own the Asterisk code, but it possesses enough of a copyright interest in the code, as well as licenses from all contributors, in order to be able to make that exception. There is a way for Howlertech to keep their G.729 licensing code under wraps, run against Asterisk, and comply with the GPL, but they haven't done it. They'd have to separate out the interface code from the codec code, develop a pipe interface (or shared memory interface, i.e. SYSVSHM) between the two, run their binary blob in a separate process, and distribute the source for their interface code to Asterisk, along with the binary blob, enough for someone to be able to compile the module themselves. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx database
I have tried all that. I just can't trace the value. this maybe the wrong list. I just thought someone might know On Thu, Sep 17, 2009 at 5:39 PM, cov...@ccs.covici.com wrote: Well, why not disable it from the GUI and see what changes -- this is sort of the wrong list, but maybe someone knows more fully. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP Users Conference Friday: Andy Abramson of VoIP Watch
Hi, This week we are pleased to welcome Andy Abramson (http://andyabramson.blogs.com/voipwatch/) as our guest. Andy is one of the most avid observers of the world of VoIP, from Asterisk and its variations to all kinds of ramifications of VoIP. I'm sure we'll pass a lot of the VoIP News in review tomorrow at the usual time and place: Friday 18 Sept at 12 Noon EDT, your local time: http://permatime.com/America/New_York/2009-09-18/12:00/VoIP_Users_conference Be on IRC #voip-users-conference on Freenode.net if you can sip:7463#2262...@proxy.ideasip.com for g711 sip:200...@login.zipdx.com for g722 http://VUC.me for general information Join us any Friday and come to meet many of the VUC regulars at Astricon! /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
Close enough. Digium doesn't own the Asterisk code, but it possesses enough of a copyright interest in the code, as well as licenses from all contributors, in order to be able to make that exception. Ah, yeah, I stand corrected. I should have not used own. For the casual reader, the clarification means developers contributing to Asterisk still own the code, but the disclaimer signed by them gives Digium enough rights to make the exception. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 RTP Transmission error of packet
Nobody on this ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi Sent: September-16-09 7:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323 RTP Transmission error of packet Using H323 to reach another h323 switch, I have no audio and the following error: [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21282 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21283 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21284 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21285 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21286 to XXX.XXX.XXX.XXX:6064: Invalid argument Can you please tell me what I`m missing I`m doing a quick dial like Dial(h323/1514...@xxx.xxx.xxx.xxx) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
Jared Smith jsm...@digium.com writes: Again, the emphasis on the dCAP exam is real-world knowledge of how to build a simple small-business PBX with Asterisk. If you've used Asterisk in a professional capacity, it should be very straightforward to pass the practical portion of the exam. I believe I can reveal this much without causing any problems for Digium: Be sure you have tried to configure a Polycom phone and an analog DAHDI card. Wasting 30 minutes on those two things makes passing the exam slightly more challenging... /Benny (whose only experience with analog DAHDI so far has been that dCAP exam) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
Benny Amorsen wrote: Jared Smith jsm...@digium.com writes: Not that I would ever consider taking an exam like that, but I have been using/configuring asterisk since nearly the beginning of this mailing list, and I have never touched dahdi or polycom. Someone should still be able to pass an exam without knowing about specific hardware where there is more than one alternative to use in real configurations. Again, the emphasis on the dCAP exam is real-world knowledge of how to build a simple small-business PBX with Asterisk. If you've used Asterisk in a professional capacity, it should be very straightforward to pass the practical portion of the exam. I believe I can reveal this much without causing any problems for Digium: Be sure you have tried to configure a Polycom phone and an analog DAHDI card. Wasting 30 minutes on those two things makes passing the exam slightly more challenging... /Benny (whose only experience with analog DAHDI so far has been that dCAP exam) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
Since Digium's contribution to Asterisk (hardware-wise) is Analog DAHDI cards, this makes sense (to me). I suppose they could make a DCAP exam that just used SIP trunks and softphones, but then that would just be GCAP (Generic Certified Asterisk Professional) or SCAP (SIP Certified Asterisk Professional). Just my .02. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: Thursday, September 17, 2009 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dCAP Exam Benny Amorsen wrote: Jared Smith jsm...@digium.com writes: Not that I would ever consider taking an exam like that, but I have been using/configuring asterisk since nearly the beginning of this mailing list, and I have never touched dahdi or polycom. Someone should still be able to pass an exam without knowing about specific hardware where there is more than one alternative to use in real configurations. Again, the emphasis on the dCAP exam is real-world knowledge of how to build a simple small-business PBX with Asterisk. If you've used Asterisk in a professional capacity, it should be very straightforward to pass the practical portion of the exam. I believe I can reveal this much without causing any problems for Digium: Be sure you have tried to configure a Polycom phone and an analog DAHDI card. Wasting 30 minutes on those two things makes passing the exam slightly more challenging... /Benny (whose only experience with analog DAHDI so far has been that dCAP exam) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DeadAgi
Hi people, I have the following dialplan:
[context]
exten = s,1,Noop(Start)
...
exten = h,1,Noop(Ending)
exten = h,n,DEADAGI(finconf.php,${ARG1},${ARG2})
When it is running, the asterisk gives the following error:
-- Launched AGI Script /var/lib/asterisk/agi-bin/finconf.php
== finconf.php|800|: Failed to execute
'/var/lib/asterisk/agi-bin/finconf.php': No such file or directory
But the file is there. The command ls -l returns:
-rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php
Why does it return the error?
Thanks,
Anahi Ludueña
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Re: [asterisk-users] DeadAgi
Either the file is missing something like #!/usr/bin/php or there is an
error in what the file is trying to access
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Thursday, September 17, 2009 2:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DeadAgi
Hi people, I have the following dialplan:
[context]
exten = s,1,Noop(Start)
...
exten = h,1,Noop(Ending)
exten = h,n,DEADAGI(finconf.php,${ARG1},${ARG2})
When it is running, the asterisk gives the following error:
-- Launched AGI Script /var/lib/asterisk/agi-bin/finconf.php
== finconf.php|800|: Failed to execute
'/var/lib/asterisk/agi-bin/finconf.php': No such file or directory
But the file is there. The command ls -l returns:
-rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php
Why does it return the error?
Thanks,
_
Anahi Ludueña
_
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http://download.live.com
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Re: [asterisk-users] DeadAgi
1) does the file exist
2) is it chmod'd to 755 (not sure if this matters though)
3) do you have something like #!/usr/bin/php at the start of the php file?
Cheers
Geraint
2009/9/17 Anahi Ludueña a_ludu...@hotmail.com
Hi people, I have the following dialplan:
[context]
exten = s,1,Noop(Start)
...
exten = h,1,Noop(Ending)
exten = h,n,DEADAGI(finconf.php,${ARG1},${ARG2})
When it is running, the asterisk gives the following error:
-- Launched AGI Script /var/lib/asterisk/agi-bin/finconf.php
== finconf.php|800|: Failed to execute
'/var/lib/asterisk/agi-bin/finconf.php': No such file or directory
But the file is there. The command ls -l returns:
*-rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php*
Why does it return the error?
Thanks,
*
--
*
*Anahi Ludueña*
--
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Messengerhttp://download.live.com
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Re: [asterisk-users] DeadAgi
On Thursday 17 September 2009 15:06:28 Geraint Lee wrote: 1) does the file exist 2) is it chmod'd to 755 (not sure if this matters though) 3) do you have something like #!/usr/bin/php at the start of the php file? 4) Is the file in MS-DOS format (i.e. do you have \r\n at the end of every line, instead of only \n)? That invisible character (\r) will prevent the file from executing, as Unix is looking for a file on the filesystem named /usr/bin/php\r, and that file probably doesn't exist. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] requirecalltoken and Realtime
Aloha, I too am running into a similar problem. I have version 1.6.1.5 that works fine, however 1.6.1.6 does not. I have both requirecalltoken=no calltokenignore=xxx.xxx.xxx.xxx in iax.conf, but I am still getting this error message. [Sep 17 09:54:23] ERROR[32335]: chan_iax2.c:4529 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address xxx.xxx.xxx.xxx in the calltokenignore list or setting user (null) requirecalltoken=no Thanks in advance! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DeadAgi
Thanks for the answers! The file didn't have the first line! #!/usr/bin/phpBye! Anahi Ludueña From: tles...@digium.com To: asterisk-users@lists.digium.com Date: Thu, 17 Sep 2009 15:59:21 -0500 Subject: Re: [asterisk-users] DeadAgi On Thursday 17 September 2009 15:06:28 Geraint Lee wrote: 1) does the file exist 2) is it chmod'd to 755 (not sure if this matters though) 3) do you have something like #!/usr/bin/php at the start of the php file? 4) Is the file in MS-DOS format (i.e. do you have \r\n at the end of every line, instead of only \n)? That invisible character (\r) will prevent the file from executing, as Unix is looking for a file on the filesystem named /usr/bin/php\r, and that file probably doesn't exist. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Nuevo Windows Live, un mundo lleno de posibilidades. Descúbrelo. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DeadAgi
On Thu, 17 Sep 2009, Anahi Ludue?a wrote: Thanks for the answers! The file didn't have the first line! #!/usr/bin/php Glad you found the answer. However... The command ls -l returns: -rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php Having an executable with 777 permissions is a very bad idea. Think about somebody (or some program) executing something like: echo rm -f -r /whatever-they-want \ /var/lib/asterisk/agi-bin/finconf.php -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Records for MeetMe
Hello - I am fairly new to Asterisk, but we have a fully operational system with very few hiccups. Much of that is because of this list. Thanks. My question is this - We have assigned MeetMe conference IDs to all of our employees. We then setup a TN to accommodate the MeetMe() app. Everything works fine. In fact, it works great. However, I can't seem to figure out a good way to log which conference ID that is being used. If I just call MeetMe(,Ms) in the dial plan, I get the source and destinations but no indication of what conference they joined. Ideally, I would be able to jam the conference ID being used into the account code or user field CDR fields. Thanks for any help! -Andy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone having issues with 1.6.1.6 res_snmp?
I am working on updating to 1.6.1.6 and if I have res_snmp.so auto-loading on CentOS 5.3 Asterisk Seg faults every time. I can load the module manually after the initial startup. I am starting to dig into it further and will open a ticket, just wanted to see if anyone else knew of any issues off hand, or could reproduce it. Thanks -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
On 18/09/09 7:12 AM, jon pounder wrote: Benny Amorsen wrote: Jared Smithjsm...@digium.com writes: Not that I would ever consider taking an exam like that, but I have been using/configuring asterisk since nearly the beginning of this mailing list, and I have never touched dahdi or polycom. Someone should still be able to pass an exam without knowing about specific hardware where there is more than one alternative to use in real configurations. To be fair, I'm about the same, but have used both DAHDI analogue lines and had to use them with Polycom phones. It's not really that difficult as long as you understand the tftp provisioning for the phones. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom auto-install asterisk using ks.cfg
Hi guys, Anyone done this with CentOS and asterisk 1.4? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] limit concurrent calls on trunk supporting multiple DID
At 7:16 AM on 17 Sep 2009, Patrick wrote:
I've one SIP trunk that support multiple DID. Only the trunk is
documented in sip.conf (called DID is taken from the sip-header in
real time).
I would like to limit the number of simultaneous calls on each DID. Is
there a way to achieve this ?
I think you could use GROUP() and GROUPCOUNT() for that. I do that for
Queue calls currently, so each agent only gets one call at a time. It
would go something like this (entirely untested):
[incoming]
exten = _X.,1,Set(DID=${EXTEN})
exten = _X.,n,GotoIf($[GROUP_COUNT(${DID})=0]?accept)
exten = _X.,n,Busy()
exten = _X.,n(accept),Set(GROUP()=${DID})
; Now let the call through as usual...
exten = _X.,n,Goto(mainmenu,s,1)
That puts each call into a group named by the DID, and returns Busy
if there is another call on the same DID.
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[asterisk-users] DAHDI Caller ID problem
Aloha,
I'm finishing up the final touches on this install, and have run into an
odd problem.
I can't seem to get Caller ID on the analog phone lines working. It's a
Digium AEX 410 card.
I have Verbose set and a line to print the CID:
I have usecallerid=yes and callerid=asreceived set in both chan_dahdi.conf
and users.conf
[analog]
include=default
exten = s,1,Verbose(passed id is ${CALLERID(num)})
exten = s,2,Answer
exten = s,3,Dial(SIP/100,,)
And this is what I'm getting.
*CLI core set verbose 10
Verbosity was 1 and is now 10
-- Starting simple switch on 'DAHDI/1-1'
[Sep 17 14:44:05] NOTICE[15308]: chan_dahdi.c:7542 ss_thread: Got event 18
(Ring Begin)...
[Sep 17 14:44:07] NOTICE[15308]: chan_dahdi.c:7542 ss_thread: Got event 2
(Ring/Answered)...
[Sep 17 14:44:07] NOTICE[15308]: chan_dahdi.c:7706 ss_thread: MWI: Channel
1 message waiting!
-- Executing [...@analog:1] Verbose(DAHDI/1-1, passed id is ) in new
stack
passed id is
-- Executing [...@analog:2] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@analog:3] Dial(DAHDI/1-1, SIP/100,,) in new stack
== Using SIP RTP CoS mark 5
-- Called 100
-- SIP/100-b6a22338 is ringing
-- SIP/100-b6a22338 answered DAHDI/1-1
== Spawn extension (analog, s, 3) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
I'm also getting these errors:
[Sep 17 14:01:06] ERROR[14462]: callerid.c:562 callerid_feed: No start bit
found in fsk data.
[Sep 17 14:01:06] WARNING[14462]: chan_dahdi.c:7582 ss_thread: CallerID
feed failed: Success
[Sep 17 14:01:06] WARNING[14462]: chan_dahdi.c:7686 ss_thread: CallerID
returned with error on channel 'DAHDI/1-1'
I have tried calleridsignal=dtmf ring, as well as calleridstart=ring
polarity. No love.
I searched on google for info, but nothing I found had a solution for my
problem.
I know that there's something I missing, but I can't seem to figure it
out. Can you all help me?
Thanks in advance!
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Re: [asterisk-users] Custom auto-install asterisk using ks.cfg
I have with CentOS 5.3 and custom 1.6.1.6 RPMs. If you use RPMs for the installation of Asterisk then it's really easy. As for the Kickstart, if you haven't used it much here I did a quick write-up with example script here: http://thurmantech.com/node/3 Either use RPMs and add them to the packages section, or download the tar.gz file in the post script, and auto-compile. However, auto-compile might have different results on different systems (hence why I use custom RPMs) -Jonathan On Thu, Sep 17, 2009 at 4:27 PM, Neeraj Chand neeraj.ch...@ocis.com.au wrote: Hi guys, Anyone done this with CentOS and asterisk 1.4? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 404 for SUBSCRIBE
Hi, All, I always get 404 respond when I send SUBSCRIBE to asterisk. Does anybody know why? Message flow is as follows: SUBSCRIBE sip:1...@192.168.1.32:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:5060;rport;branch=z9hG4bKPjdf42536ef66447529e3da381f3556c1b Max-Forwards: 70 From: 1003 sip:1...@192.168.1.32;tag=b23ca2205f964a4b951ddb90326f544a To: sip:1...@192.168.1.32 Contact: sip:1...@192.168.1.36:5060 Call-ID: 25a03edb881c4b8582baccea451078bc CSeq: 18467 SUBSCRIBE Event: dialog Expires: 300 Accept: application/dialog-info+xml Allow-Events: refer, dialog Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.36:5060;branch=z9hG4bKPjdf42536ef66447529e3da381f3556c1b;received=192.168.1.36;rport=5060 From: 1003 sip:1...@192.168.1.32;tag=b23ca2205f964a4b951ddb90326f544a To: sip:1...@192.168.1.32;tag=as29c6808b Call-ID: 25a03edb881c4b8582baccea451078bc CSeq: 18467 SUBSCRIBE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=24c2056c Content-Length: 0 SUBSCRIBE sip:1...@192.168.1.32:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:5060;rport;branch=z9hG4bKPjb8a9f275986c45d596f56e747aac0e48 Max-Forwards: 70 From: 1003 sip:1...@192.168.1.32;tag=b23ca2205f964a4b951ddb90326f544a To: sip:1...@192.168.1.32 Contact: sip:1...@192.168.1.36:5060 Call-ID: 25a03edb881c4b8582baccea451078bc CSeq: 18468 SUBSCRIBE Event: dialog Expires: 300 Accept: application/dialog-info+xml Allow-Events: refer, dialog Authorization: Digest username=1003, realm=asterisk, nonce=24c2056c, uri=sip:1...@192.168.1.32:5060, response=0c3c8ae04133a14b913f906ee2ca9c02, algorithm=MD5 Content-Length: 0 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.36:5060;branch=z9hG4bKPjb8a9f275986c45d596f56e747aac0e48;received=192.168.1.36;rport=5060 From: 1003 sip:1...@192.168.1.32;tag=b23ca2205f964a4b951ddb90326f544a To: sip:1...@192.168.1.32;tag=as29c6808b Call-ID: 25a03edb881c4b8582baccea451078bc CSeq: 18468 SUBSCRIBE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
