Re: [Asterisk-Users] Best software SIP client
if you're on windoze, msn 4.7 works fine. v5 and v6 doesn't, though roy On Mon, 2003-07-21 at 12:30, Stuart Hirst wrote: Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but is not the most user friendly and caused odd errors on the Asterisk console. What I am looking for is a software SIP client that is simple for users to use ( they don't have to understand SIP ) and that works reliably. Rgds, Stuart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] File chan_h323.c, Line 875
Hello, Anybody experience this error: ERROR[237594]: File chan_h323.c, Line 875 (create_connection): Unable to allocate private structure, this is very bad. the call still get through, but both party cannot hear each other Pls Help. Foong - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 25, 2003 1:47 PM Subject: Re: [Asterisk-Users] Voicemail() problems - Long pause afterincoming message recording ended. On Thu, 2003-07-24 at 22:38, [EMAIL PROTECTED] wrote: I'm having the following problem: I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card) to access voicemail. After dialing the appropriate extension I get voicemail, am presented with the user's unavailable message, and can leave a message normally. The problem comes when I press # to end the recording, at which point I am told Your message has been saved. Then there is a long lag of about 20 seconds of silence, during which Asterisk does not respond to DTMF at all, before I am finally dropped back into the priority list for the extension, which in this case is a simple Goodbye hangup. Any idea why this long lag after message-recording termination is happening? I'd like Asterisk to hang up immediately after the incoming caller terminates their VM recording. Check your mail settings, also whether your DNS is working fast. This is all probably due to the time to get the vm out the app and into the mail. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX and Call format
Hi, What the Call Format means in IAX? For example: -- Format for call is 2 It is codec related? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] time and date stamp in voicemail
Hi, This page does not exist... Thanks, Dan - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 25, 2003 8:38 AM Subject: RE: [Asterisk-Users] time and date stamp in voicemail Try looking drunkencoder.com/asterisk On Thu, 2003-07-24 at 22:16, Andy Hester wrote: I have searched and not located this patch...is there a specific place that I need to look, or a specific file name? Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Thursday, July 24, 2003 6:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] time and date stamp in voicemail Hi, I see that there's been some very light discussion on having a standard time and date stamp in VM. How can I implement it today? (About to offer a system to a customer but they need the stamp to tell when people called.) Thanks, -- Steve __ This sig is pending approval Tilghman Lesher had a well-written patch he posted to the list a few weeks ago for Voicemail, which I've been using without difficulty. He has said he's going to work on Voicemail2, so I am hoping to see that soon, and then integration into the main CVS tree after some testing. Not only does the patch handle generic time and date stamps, but it allows customizable timezones, announcement strings, and uses standard UNIX-ish time code macros. Very slick, and really necessary for voicemail systems that happen to have users in multiple timezones. If you are looking for words to match Tilghman's patch, see the phrases I have submitted (and donated by a generous grant by VoicePulse) as public domain, recorded by Allison Smith: http://www.loligo.com/asterisk/sounds/ JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Integrating cell phone into Asterisk Extension..
Granted it takes some pre-thought, but why
not just use the *72/*73 codes to forward
when you are out of the office?
This worked out pretty well, I got to meet lots of the
other officemembers' family and friends! Ha ha!
Seriously, I think Asterisk can do better, just need
to figure out where to do it *from*. Seems like a
'forward-timeout' app could be pretty useful. Dial a
number, wait for a '#' on channels where pickup isn't
detected, read out the callerID, put the caller
through on another '#' or (for example) through to
voicemail on '*', hangup or timeout.
Looks like AGI won't quite do the trick. I've been
looking over Cam's HOWTO:
http://home.cogeco.ca/~camstuff/agi.html
AGI seems to be missing a way to dial out (Answer
won't do it since you can't specify a channel, and
even we could, there's no convenient way of sending
DTMF), and once that were done, I'm unclear how you'd
pipe the two channels together (sorry for the UNIX
shell terminology!).
It's looking at this point like the best bet is to
modify the Dial application, or write a new one. Does
that sound reasonable to anyone? As in, am I barking
up the right tree yet? I'd hate to spend a couple of
days on a problem that someone else could write in two
lines of extensions.conf syntax on a 2x4 before
whopping me in the face with it.
[trycell]
; Add a new option 'p' to indicate a dumb POTS line
; that can't do CallerID or answer detection
; ${POTSLINES} could be FXO devices, eg. Zap/g1
exten = s,1,Dial(${POTSLINES}/${MYCELL},20,Trp)
exten = s,2,Voicemail,${MYEXTEN}
BTW, what ARE the semantics of this?:
; ${TDM400P} are the ports on an FXS card, maybe
; TDM400P = Zap/3Zap/4
exten = s,1,Dial(${TDM400P}${POTSLINES}/${MYCELL})
John
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Wed, 2003-07-23 at 02:26, John Morris wrote:
This is a thread from Apr. 15 about how to make
your
cell phone ring simultaneously with your other
extensions. The solution was:
Just use Dial,Zap/${DESK}Zap/g1/${CELL}||t
I've tried this, and this is the behavior I see:
1) ${DESK} extensions ring ONCE only (in my case,
I
have the TDM400P with analog phones as extensions)
2) ${DESK} extensions are hung up after just a
few
seconds
3) ${CELL} is dialed
Any idea why this would happen?
If you have a tdm400P I assume you probably have an
X100P also. If you
are dialing outside with an analog circuit you don't
have definitive
knowledge of them the other side is answered. With
out this knowledge,
your cell phone answered the line first and the
desk phones loose the
opportunity to answer.
SO, since this didn't work, I figured I'd ring
${DESK}
for 15 seconds first; if there was no pickup, I'd
ring
${CELL} second; and if no pickup there, forward to
voicemail:
[john]
; my very own context!
exten = s,1,Dial(${DESK},20)
exten = t,1,Goto(john-cell,s,1)
Might want to change this last line from t to s,2.
If I remember
correctly, your dial will time out and then go into
limbo till
DigitTimeout, then go to t.
[john-cell]
; call my cell if I don't answer
exten = s,1,Dial(Zap/g1/${CELL},20)
; go to voicemail if I don't pick up the cell
exten = t,1,Voicemail,1002
exten = t,2,hangup
This doesn't do what I need, either. If I don't
pick
up ${DESK}, then ${CELL} is dialed. Problem is,
even
if I answer ${CELL}, asterisk doesn't register
that I
picked up, and after 20 seconds it hangs up on
${CELL}
and goes to voicemail. I'm using two X100Ps for
FXO
devices, by the way.
Do you have an AbsoluteTimeout set somewhere? BTW,
your timeout
shouldn't work for the cell phone since the X100P
won't know when you
answered the line or not, it will just assume
answered.
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] Integrating cell phone into Asterisk Extension..
Granted it takes some pre-thought, but why
not just use the *72/*73 codes to forward
when you are out of the office?
This worked out pretty well, I got to meet lots of the
other officemembers' family and friends! Ha ha!
Seriously, I think Asterisk can do better, just need
to figure out where to do it *from*. Seems like a
'forward-timeout' app could be pretty useful. Dial a
number, wait for a '#' on channels where pickup isn't
detected, read out the callerID, put the caller
through on another '#' or (for example) through to
voicemail on '*', hangup or timeout.
Looks like AGI won't quite do the trick. I've been
looking over Cam's HOWTO:
http://home.cogeco.ca/~camstuff/agi.html
AGI seems to be missing a way to dial out (Answer
won't do it since you can't specify a channel, and
even we could, there's no convenient way of sending
DTMF), and once that were done, I'm unclear how you'd
pipe the two channels together (sorry for the UNIX
shell terminology!).
It's looking at this point like the best bet is to
modify the Dial application, or write a new one. Does
that sound reasonable to anyone? As in, am I barking
up the right tree yet? I'd hate to spend a couple of
days on a problem that someone else could write in two
lines of extensions.conf syntax on a 2x4 before
whopping me in the face with it.
[trycell]
; Add a new option 'p' to indicate a dumb POTS line
; that can't do CallerID or answer detection
; ${POTSLINES} could be FXO devices, eg. Zap/g1
exten = s,1,Dial(${POTSLINES}/${MYCELL},20,Trp)
exten = s,2,Voicemail,${MYEXTEN}
BTW, what ARE the semantics of this?:
; ${TDM400P} are the ports on an FXS card, maybe
; TDM400P = Zap/3Zap/4
exten = s,1,Dial(${TDM400P}${POTSLINES}/${MYCELL})
John
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Wed, 2003-07-23 at 02:26, John Morris wrote:
This is a thread from Apr. 15 about how to make
your
cell phone ring simultaneously with your other
extensions. The solution was:
Just use Dial,Zap/${DESK}Zap/g1/${CELL}||t
I've tried this, and this is the behavior I see:
1) ${DESK} extensions ring ONCE only (in my case,
I
have the TDM400P with analog phones as extensions)
2) ${DESK} extensions are hung up after just a
few
seconds
3) ${CELL} is dialed
Any idea why this would happen?
If you have a tdm400P I assume you probably have an
X100P also. If you
are dialing outside with an analog circuit you don't
have definitive
knowledge of them the other side is answered. With
out this knowledge,
your cell phone answered the line first and the
desk phones loose the
opportunity to answer.
SO, since this didn't work, I figured I'd ring
${DESK}
for 15 seconds first; if there was no pickup, I'd
ring
${CELL} second; and if no pickup there, forward to
voicemail:
[john]
; my very own context!
exten = s,1,Dial(${DESK},20)
exten = t,1,Goto(john-cell,s,1)
Might want to change this last line from t to s,2.
If I remember
correctly, your dial will time out and then go into
limbo till
DigitTimeout, then go to t.
[john-cell]
; call my cell if I don't answer
exten = s,1,Dial(Zap/g1/${CELL},20)
; go to voicemail if I don't pick up the cell
exten = t,1,Voicemail,1002
exten = t,2,hangup
This doesn't do what I need, either. If I don't
pick
up ${DESK}, then ${CELL} is dialed. Problem is,
even
if I answer ${CELL}, asterisk doesn't register
that I
picked up, and after 20 seconds it hangs up on
${CELL}
and goes to voicemail. I'm using two X100Ps for
FXO
devices, by the way.
Do you have an AbsoluteTimeout set somewhere? BTW,
your timeout
shouldn't work for the cell phone since the X100P
won't know when you
answered the line or not, it will just assume
answered.
--
Steven Critchfield [EMAIL PROTECTED]
___
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Re: [Asterisk-Users] Cisco's CallManager and * (was: Cisco 7960g)(fwd)
On Fri, 25 Jul 2003, Kelvin Chua wrote: yes, i agree, we never really felt the need to use unity, *'s vm is functionally ok with callmanager (except for the message waiting indication, or is there?) can *'s vm send a MWI to the callmanager? Not yet. However, CCM supports MWI notifications from foreign voicemail systems via a serial (yes, those old RS-232) interface, so it _should_ be possible to add that. I bet there's some other (undocumented?) interface as well... Anyhow, it'll be some time before I'll be looking into MWI. And even if that doesn't work, you can always push a message to the Display that says You have n new messages along with a softkey to read them... Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail enhancements
On Thu, 24 Jul 2003, Daryl Jones wrote: Brad's recent list of enhancements look good, but I haven't looked at the code yet. If the code looks good, I hope it will be committed to the project CVS. Thanks for the vote of confidence but I fear it's premature. Hopefully someday I can make it suitable for CVS though. Here's a partial list of enhancements that I would like to see in Comedian Mail. I am probably interested in helping to fund the enhancement of Asterisk voicemail. Is anyone else interested? -Address message to multiple recipients -Forward message to multiple recipients -Recipient groups maintained by a user-accessible web page -Scheduled future deliver -Say digits of calling number when requested -Reminder paging notification until message is retrieved Those would all be cool. Calling party number I have implemented as part of message envelope, where I have tried a somewhat dubious method to play the recorded name if the call originated from another system user. I'm definitely interested Comedian Mail myself at the moment (to the extent that I'm qualified to muddle with it). My own to-do list still includes a few items: - flagging of messages as urgent, private, etc - handling urgent, private, etc messages appropriately (still pondering a sensible way to give urgent messages priority during playback) - mailbox options for individual users to specify a couple of numbers (other than 0 for operator) to which a caller could transfer him/herself - a secretarial password that would allow listening to message headers for new messages and nothing else Brad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] URGENT! brandly new Wildcard E400P for sale at $1000
Title: Message Hi, Do you still have it for sale ? Abdul Hakeem -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kalin DikovSent: 21 July 2003 20:14To: [EMAIL PROTECTED]Subject: [Asterisk-Users] URGENT! brandly new Wildcard E400P for sale at $1000Importance: High A brandly new E400P 128 channel PRI, you can find more information on the Digium's site. The card was not used before, I sell it cause our company just don't need, we use Cisco AS5300. I can offer you the PCI PRI at $940 if bought in the next 2 days, I can send it to you via FedEx or DHL express. Please email me for more details! Regards, Kalin - IPPN Networks LTD http://www.ippn.net technical manager mobile: +35998804462-
[Asterisk-Users] Asterisk /SIP .. nat
Hi. Is it posible to put a Asterisk server behind a NAT-firewall, and let it be reachable from the Internet.. Like this: asterisk - nat - inet - SIP or am I forced to have the Asterisk box connected "directly" on the internet?.. Than
[Asterisk-Users] Dialogic hardware
Hi all ! What is the current status of the Dialogic channel driver ? Is it available ? Is it commercial ?? Any info ? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialogic hardware
I asked the same question a couple of weeks ago and was told by Digium that its not commercially available yet but the source code is available under NDA with Digium. I'll dig out my contact and send off-list ... Adam -Original Message- From: Marcel Prisi [mailto:[EMAIL PROTECTED] Sent: 25 July 2003 11:30 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dialogic hardware Hi all ! What is the current status of the Dialogic channel driver ? Is it available ? Is it commercial ?? Any info ? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk /SIP .. nat
It probably can be done but you would have to forward a number of ports from the NAT to the * box.. Also be carefull of double NAT situations.. ie. Asterisk--NAT--internet--NAT--UA becasue this is a real headache to get working..if you ever do.. Later Hi. Is it posible to put a Asterisk server behind a NAT-firewall, and let it be reachable from the Internet.. Like this: asterisk - nat - inet - SIP or am I forced to have the Asterisk box connected directly on the internet?.. Than -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best software SIP client
I am trying V2 now but having problems. I am having the same issues I had with build 1016 in that the call is set-up OK but no voice path in either direction. I guess this is one of the setting in V2 1016 but I have not worked out which one yet. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Woodhead Sent: 25 July 2003 10:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Best software SIP client Hi Roy, X-Lite was the best we found for user friendliness but we had quality problems which I've mentioned on here several times. Have you tried v2 though? We just have and all of the problems we had with the earlier version have been overcome. Well worth a look! W - Original Message - From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 25, 2003 8:38 AM Subject: Re: [Asterisk-Users] Best software SIP client if you're on windoze, msn 4.7 works fine. v5 and v6 doesn't, though roy On Mon, 2003-07-21 at 12:30, Stuart Hirst wrote: Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but is not the most user friendly and caused odd errors on the Asterisk console. What I am looking for is a software SIP client that is simple for users to use ( they don't have to understand SIP ) and that works reliably. Rgds, Stuart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] go on in context after the destination channel hung up?
Hi all, is it possible to go on in the context after the dest channel hung up? For example: exten = 111,1,Dial,Zap/4 If the originating channel is connected to Zap/4 and the destination channel (Zap/4) hangs up, both channels will be destroyed. Is there any option or whatever for preventing the hangup for the originating channel and go on in the current context ? like : exten = 111,1,Dial,Zap/4 (after Zap/4 has hung up) exten = 111,2,whatever ... Any ideas ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuration sample for isdn4linux?
Hi, we want to use asterisk as a replacement for our current pbx but before changeing we want to make some tests. Therefor I have a PC with an AVM-A1 ISDN4Linux hardware and inside the network a cisco 7960 Ip Phone (SIP) and two TIPTEL innovaphone 200 (H.323) and some softpgones like kphone, netmeeting, ... To learn something about the configuration of asterisk I want to create a small config where all my clients can call each other AND ca use the two B-Channels of my isdn4linux-card. I read the asterisk handbook and I tried hard to use the installed config files but I have real problems to understand the examples in the handbook or the configs because they are for all reasons. Does anyone have such a configuration in use and can send me the configs? This would be very great! I hope to learn more by using a working config and try to get it to work with my environment. Thanx, Holger -- # ## ## Holger Wirtz Phone : (+49 30) 884299-40 ## ## ## ### ## DFN-Verein Fax : (+49 30) 884299-70 ## ## ## Anhalter Str. 1 E-Mail: [EMAIL PROTECTED] ## ## ## ## ### 10963 Berlin # ## ## ## GERMANY WWW : http://www.dfn.de GPG-Fingerprint: ABFA 1F51 DD8D 503C 85DC 0C51 E961 79E2 6685 9BCF ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SetLanguage application doesn;t seem to work in latest Asterisk
Hello everybody, I have installed Asterisk from CVS (18/07/2003) and although everything works fine, SetLanguage application doesn't seem to work. As it used to work with previous version I wonder if I am missing something here. The relevant line in extensions.conf is: exten = 3,1,SetLanguage(gr) In the directory where Asterisk sounf files reside, I have installed files of the type file-gr.gsm. However, although the SetLanguage(gr) application is executed, only the plain file.gsm files are played. Note that I haven't setup gr as a language in any other configuration file. Any clues are welcome. Thanks Anna Panagidou Technology Department Hellas On Line Agiou Konstantinou 59-61 15124, Maroussi Tel. no: (+30210) 8762309 E-mail address: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best software SIP client
Does anyone have X-Lite v2 Build 1047 (the new one) working ? Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Hirst Sent: 25 July 2003 10:42 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client I am trying V2 now but having problems. I am having the same issues I had with build 1016 in that the call is set-up OK but no voice path in either direction. I guess this is one of the setting in V2 1016 but I have not worked out which one yet. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Woodhead Sent: 25 July 2003 10:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Best software SIP client Hi Roy, X-Lite was the best we found for user friendliness but we had quality problems which I've mentioned on here several times. Have you tried v2 though? We just have and all of the problems we had with the earlier version have been overcome. Well worth a look! W - Original Message - From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 25, 2003 8:38 AM Subject: Re: [Asterisk-Users] Best software SIP client if you're on windoze, msn 4.7 works fine. v5 and v6 doesn't, though roy On Mon, 2003-07-21 at 12:30, Stuart Hirst wrote: Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but is not the most user friendly and caused odd errors on the Asterisk console. What I am looking for is a software SIP client that is simple for users to use ( they don't have to understand SIP ) and that works reliably. Rgds, Stuart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a stand alone voice mail server
On Thu, 24 Jul 2003, Jeremy McNamara wrote: Siggi Langauf wrote: Are you running CCM 3.3(2), too? No idea, I avoid dealing with CCM at all I fought tooth and nail to stop them from wasting money on it, but they wouldn't listen to me. Same thing here: they're probably going to pour Millions of Euros into the Cisco dump during the next year. Unless I manage convince them to use *, of course ;-) But now I'm getting confused: you're _not_ using CCM?? OTOH, your scenario doesn't look like it's involving any Skinny phones, so maybe your CCM doesn't setup streams for foreign (IP phones') IP addresses, which would explain why you're not hitting this bug... I don't know how u can get away with not using Skinny on CCM. We use Skinny phones into H.323 trunks for voicemail. Instead of wasting more money on Unity, I talked them into Asterisk voicemail. Even without all the fancy integration, they are still happy. So you _do_ have CCM, and Cisco's Skinny phones, and this setup does actually work for you: Cisco 7960 --(Skinny)-- CCM ^ ^ | | RTP H.323 call setup | | +--- * chan_h323 -+ | *--- app_voicemail2 That RTP connection on the left-hand side was misdirected to CCM for me, until I switched to chan_oh323... Puzzled, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7940 AS5300 codec issues/questions G.729 G.711
I've previously been using G711alaw on both the AS5300 and the phones but feel the need for a less bandwidth hungry codec for those users that are connected behind ADSL and so was investigating G.729 but .. Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940 phones I have G.729a, I'm not sure which interoperate the best with each other and so was wondering if anyone call tell me if they have similar setups with this working and if so which codec they choose for the AS5300 ... When comparing the G711alaw against G729a I did not expect to have so many breaks in the sound when using G729a and wondered if others had experienced this. I would expect G729 to be a lesser overall quality and sampling rate but not to effectively lose some speech altogether ... * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi error
hello, sometimes my capi_channel stop works - e.g. when i try to call number which does not exist ( typo error ) and i must restart asterisk. following lines appears in the log files : ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free channel on controller 1! will continue searching. ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1! will continue searching. NOTICE[393234]: File chan_capi.c, Line 1060 (capi_request): didn't find capi device with outgoing msn = 5430754. you should check your config! ERROR[311314]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1! will continue searching. ERROR[311314]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1! will continue searching. NOTICE[311314]: File chan_capi.c, Line 1060 (capi_request): didn't find capi device with outgoing msn = 5430754. you should check your config! NOTICE[311314]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type 'CAPI' does anybody know how to solve this ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi error
upgrade to 0.2.4 :) On Friday 25 July 2003 14:09, Marian Danisek wrote: hello, sometimes my capi_channel stop works - e.g. when i try to call number which does not exist ( typo error ) and i must restart asterisk. following lines appears in the log files : ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free channel on controller 1! will continue searching. ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1! will continue searching. NOTICE[393234]: File chan_capi.c, Line 1060 (capi_request): didn't find capi device with outgoing msn = 5430754. you should check your config! ERROR[311314]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1! will continue searching. ERROR[311314]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1! will continue searching. NOTICE[311314]: File chan_capi.c, Line 1060 (capi_request): didn't find capi device with outgoing msn = 5430754. you should check your config! NOTICE[311314]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type 'CAPI' does anybody know how to solve this ? regards Marian -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 2254 5070 (work) +47 9801 3356 (mobile) Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a stand alone voice mail server
Siggi Langauf wrote: No idea, I avoid dealing with CCM at all I fought tooth and nail to stop them from wasting money on it, but they wouldn't listen to me. Same thing here: they're probably going to pour Millions of Euros into the Cisco dump during the next year. Unless I manage convince them to use *, of course ;-) But now I'm getting confused: you're _not_ using CCM?? I have nothing to do with Cisco or CCM, no. (Hell no)A few of my customers wasted money on CCM and are now looking to me to pull them out of the hole they dug themselves by not listening to me in the first place. OTOH, your scenario doesn't look like it's involving any Skinny phones, so maybe your CCM doesn't setup streams for foreign (IP phones') IP addresses, which would explain why you're not hitting this bug... I don't know how u can get away with not using Skinny on CCM. We use Skinny phones into H.323 trunks for voicemail. Instead of wasting more money on Unity, I talked them into Asterisk voicemail. Even without all the fancy integration, they are still happy. So you _do_ have CCM, and Cisco's Skinny phones, and this setup does actually work for you: Cisco 7960 --(Skinny)-- CCM ^ ^ | | RTP H.323 call setup | | +--- * chan_h323 -+ | *--- app_voicemail2 That RTP connection on the left-hand side was misdirected to CCM for me, until I switched to chan_oh323... I never bothered to check out the actual RTP path, but the one customer I have implemented Asterisk in their CCM based network has no complaints, so something must be working right. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuration sample for isdn4linux?
Holger, Therefor I have a PC with an AVM-A1 ISDN4Linux hardware and inside the network a cisco 7960 Ip Phone (SIP) and two TIPTEL innovaphone 200 (H.323) and some softpgones like kphone, netmeeting, ... [..] Does anyone have such a configuration in use and can send me the configs? This would be very great! I hope to learn more by using a working config and try to get it to work with my environment. I am using an AVM card with the chan_capi and it is working quite good. What kind of problems are you experiencing? And, what OS are you running at? rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSN Messenger(4.7) Setup
Tools | Options | Accounts Choose a communication service and type your username click 'Advanced' type the name of the server and choose 'udp' press ok until you're out open regedit browse to HKEY_CURRENT_USER\Software\Microsoft\MessengerService change CorpPC2Phone to 1 play On Friday 25 July 2003 15:52, Neel Datta wrote: I have seen a number of posts saying the MSN Messenger 4.7 works as a SIP client, but I'm having trouble figuring it out. I've gotten as far as getting the dial pad, but I can't seem to find where to specify my * server or the SIP name/password. Any help is appreciated. Neel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 2254 5070 (work) +47 9801 3356 (mobile) Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best software SIP client
I am having the same probs. I get local dialing tones but no audio after the call is connected.. I got a private build from Xten and it was the same Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 gateway call lost after 74sec always
Steven Thomas wrote: Michael, my mistake - more testing confirmed that the wrapper did not update in the correct location. Asterisk was still using 0.5.3. Replaced with 0.5.4 and the call is no longer dropped. Asterisk still reports H.323 CONTROL PROTOCOL ERROR (Roundtrip Delay) but it keeps hold of the call. Yes, don't worry about it, it's just a warning, to notify you that the remote end doesn't support it. Thanks for you help. Regards, Steve. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best software SIP client
Fixed it I have audio now... uninstall everything xten makes and manually clear out all the xten/xlite stuff from the registry.. search for XtenNetwork and kill the keys. reinstall Xpro and it works... go figure Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Instant hangup on busy Zap channel.
Do 'iax2 debug' to see more.
Martin
On Fri, 25 Jul 2003, Richard Scobie wrote:
A call is placed via IAX2 from one asterisk to another, to a TDM400
channel whose extensions.conf entry is
exten = 502,1,Dial(${COLIN})
exten = 502,2,Congestion
If this channel is already busy when called, the call is instantly
hungup, without the caller hearing the congestion tone.
The log from the callers asterisk shows:
-- Executing Dial(Zap/1-1, IAX2/192.168.3.223/502|30) in new stack
-- Called 192.168.3.223/502
-- Call accepted by 192.168.3.223 (format 4)
-- Format for call is 4
-- IAX2[pbxak]/2 is circuit-busy
== Everyone is busy at this time
-- Hungup 'IAX2[pbxak]/2'
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (incoming, 502, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Is this expected behavior, or have I missed something in the configuration?
Thanks,
Richard
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RE: [Asterisk-Users] Best software SIP client
Any one able to tell me where this can be downloaded form? I only see the generic download link - http://www.xten.com/download.php but this seems to be version 1 Thanks Damian From:Dave Packham [mailto:[EMAIL PROTECTED] Sent:Fri 25/07/2003 15:38 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Fixed it I have audio now... uninstall everything xten makes and manually clear out all the xten/xlite stuff from the registry.. search for XtenNetwork and kill the keys. reinstall Xpro and it works... go figure Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
RE: [Asterisk-Users] Voicemail() problems - Long pause afterincoming message recording ended.
What Steve says... Also, check your hosts file for strange entries. This was the problem in our case. We had exactly the same symptoms with RedHat 8. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, July 25, 2003 1:47 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicemail() problems - Long pause afterincoming message recording ended. On Thu, 2003-07-24 at 22:38, [EMAIL PROTECTED] wrote: I'm having the following problem: I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card) to access voicemail. After dialing the appropriate extension I get voicemail, am presented with the user's unavailable message, and can leave a message normally. The problem comes when I press # to end the recording, at which point I am told Your message has been saved. Then there is a long lag of about 20 seconds of silence, during which Asterisk does not respond to DTMF at all, before I am finally dropped back into the priority list for the extension, which in this case is a simple Goodbye hangup. Any idea why this long lag after message-recording termination is happening? I'd like Asterisk to hang up immediately after the incoming caller terminates their VM recording. Check your mail settings, also whether your DNS is working fast. This is all probably due to the time to get the vm out the app and into the mail. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSN Messenger(4.7) Setup
Thanks Roy, I this worked! Only one thing I can't seem to do- If I have a password set in my sip.conf as in the 'secret' key, I can't get the msn client to authenticate properly. (And yes, I'm typing the exact same word I have in secret =) Neel Tools | Options | Accounts Choose a communication service and type your username click 'Advanced' type the name of the server and choose 'udp' press ok until you're out open regedit browse to HKEY_CURRENT_USER\Software\Microsoft\MessengerService change CorpPC2Phone to 1 play ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best software SIP client
Dave I tried this and I still have the same problem. I am using X-Lite though and not X-Pro. The SIP registration is fine but still no audio. If anyone has X-Lite either 1016 or 1047 (v2) working, please could you let me know and maybe email your registry settings for the app. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Packham Sent: 25 July 2003 15:38 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Fixed it I have audio now... uninstall everything xten makes and manually clear out all the xten/xlite stuff from the registry.. search for XtenNetwork and kill the keys. reinstall Xpro and it works... go figure Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best software SIP client
Try http://brands.xten.net/x-lite/download/X-Lite_Install.exe -Original Message- From: Asterisk Maillist [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Maillist Sent: 25 July 2003 16:10 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Any one able to tell me where this can be downloaded form? I only see the generic download link - http://www.xten.com/download.php but this seems to be version 1 Thanks Damian _ From:Dave Packham [mailto:[EMAIL PROTECTED] Sent:Fri 25/07/2003 15:38 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Fixed it I have audio now... uninstall everything xten makes and manually clear out all the xten/xlite stuff from the registry.. search for XtenNetwork and kill the keys. reinstall Xpro and it works... go figure Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users attachment: winmail.dat
RE: [Asterisk-Users] Voicemail() problems - Long pause after incoming message recording ended.
I had this issue until I fixed the DNS resolver on the * box. Asterisk was attempting to deliver the mail message and having to timeout name servers, etc. Once dns was setup properly for the box, the message was delivered instantly and there was no more delay. Now a good fix would be the spawn a separate thread to deliver the message, but that was beyond my programming skill. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, July 24, 2003 10:39 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicemail() problems - Long pause after incoming message recording ended. I'm having the following problem: I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card) to access voicemail. After dialing the appropriate extension I get voicemail, am presented with the user's unavailable message, and can leave a message normally. The problem comes when I press # to end the recording, at which point I am told Your message has been saved. Then there is a long lag of about 20 seconds of silence, during which Asterisk does not respond to DTMF at all, before I am finally dropped back into the priority list for the extension, which in this case is a simple Goodbye hangup. Any idea why this long lag after message-recording termination is happening? I'd like Asterisk to hang up immediately after the incoming caller terminates their VM recording. Thanks, Sam - Here's the relevant portion of my extensions.conf file, plus output from the Asterisk CLI: [local] ; ; We start with what to do when a call first comes in. ; ;exten = s,1,Wait,1; Wait a second, just for fun exten = s,1,Answer ; Answer the line exten = s,2,DigitTimeout(2); Set Digit Timeout to 2 seconds exten = s,3,ResponseTimeout(5) ; Set Response Timeout to 5 seconds exten = s,4,BackGround(VY-ThanksForCalling); Play VY intro message (daytime) ;exten = s,5,Goto(s,4) exten = i,1,Playback(invalid) ; That's not valid, try again exten = i,2,Goto(s,4) exten = t,1,Playback(vm-goodbye) exten = t,2,Hangup exten = 2,1,Voicemail(u1000) exten = 5,1,Directory(VYStaff) exten = 7,1,ResponseTimeout(1) exten = 7,2,Voicemail(u70) -LONG PAUSE COMES HERE exten = 7,3,Playback(vm-goodbye) exten = 7,4,Hangup Asterisk CLI output: -- Set Response Timeout to 1 -- Executing VoiceMail(Zap/1-1, u70) in new stack == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm/70/unavail' -- Playing 'vm-intro' -- Playing 'beep' -- Recording to /var/spool/asterisk/vm/70/INBOX/msg0006 -- User ended message by pressing # -- Playing 'vm-msgsaved' == Parsing '/etc/asterisk/voicemail.conf': Found --LONG PAUSE COMES HERE -- Executing Playback(Zap/1-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (incoming, 7, 4) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuration sample for isdn4linux?
Hi Holger, the AVM-A1 (fritz card) has capi4linux drivers (ftp.avmd.de/cardware) which may already be included in your distro. get chan_capi at http://www.junghanns.net/asterisk/ and read the README and INSTALL file. regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Fre, 2003-07-25 um 11.55 schrieb Holger Wirtz: Hi, we want to use asterisk as a replacement for our current pbx but before changeing we want to make some tests. Therefor I have a PC with an AVM-A1 ISDN4Linux hardware and inside the network a cisco 7960 Ip Phone (SIP) and two TIPTEL innovaphone 200 (H.323) and some softpgones like kphone, netmeeting, ... To learn something about the configuration of asterisk I want to create a small config where all my clients can call each other AND ca use the two B-Channels of my isdn4linux-card. I read the asterisk handbook and I tried hard to use the installed config files but I have real problems to understand the examples in the handbook or the configs because they are for all reasons. Does anyone have such a configuration in use and can send me the configs? This would be very great! I hope to learn more by using a working config and try to get it to work with my environment. Thanx, Holger -- # ## ## Holger Wirtz Phone : (+49 30) 884299-40 ## ## ## ### ## DFN-Verein Fax : (+49 30) 884299-70 ## ## ## Anhalter Str. 1 E-Mail: [EMAIL PROTECTED] ## ## ## ## ### 10963 Berlin # ## ## ## GERMANY WWW : http://www.dfn.de GPG-Fingerprint: ABFA 1F51 DD8D 503C 85DC 0C51 E961 79E2 6685 9BCF ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best software SIP client
Stuart, X-Lite v2.0 build 1050 was just released. Try that. http://brands.xten.net/x-lite/download/X-Lite_Install.exe Cheers, Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stuart Hirst Sent: Friday, July 25, 2003 10:12 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Dave I tried this and I still have the same problem. I am using X-Lite though and not X-Pro. The SIP registration is fine but still no audio. If anyone has X-Lite either 1016 or 1047 (v2) working, please could you let me know and maybe email your registry settings for the app. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Packham Sent: 25 July 2003 15:38 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Fixed it I have audio now... uninstall everything xten makes and manually clear out all the xten/xlite stuff from the registry.. search for XtenNetwork and kill the keys. reinstall Xpro and it works... go figure Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSN Messenger(4.7) Setup
Hey Neel, On Fri, Jul 25, 2003 at 10:40:55 -0500, Neel Datta wrote: Thanks Roy, I this worked! Only one thing I can't seem to do- If I have a password set in my sip.conf as in the 'secret' key, I can't get the msn client to authenticate properly. (And yes, I'm typing the exact same word I have in secret =) Try appending @asterisk to your sign-in name in Messenger's configuration. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time and date stamp in voicemail
Dan, the page is actually http://asterisk.drunkcoder.com/patches/ . However, I didn't see the patch there. Sincerely, Andy Hester Consero -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dan Sent: Friday, July 25, 2003 2:52 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] time and date stamp in voicemail Hi, This page does not exist... Thanks, Dan - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 25, 2003 8:38 AM Subject: RE: [Asterisk-Users] time and date stamp in voicemail Try looking drunkencoder.com/asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web conf files
All I have given the * PHP web interface files to Mark to check out. hopefully he will include them into the CVS tree soon. Dave Packham ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] reconnecting
If asterisk is running as a daemon (started with no command line options), can I reconnect to it and get the vvverbose info in the console? In other words, will asterisk -rvvvc work? Sorry to ask I would just try it to see if it works, but I don't have an installation handy and have to do some remote troubleshooting soon. ThanksE+06. D. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best software SIP client
Erik, Thanks for the info. I have tried build 1050 on two different PC's and still the same symptoms. One the main machine I have been using for testing I removed the old version, cleaned the registry and installed build 1050 but still no joy. This is a Dell notebook in a docking station. The second PC is also a Dell Notebook but not as high spec and is not in a docking station. This had not had X-Lite installed before and so I just installed build 1050 with exactly the same results. I should say that build 1005 is still working fine apart from the odd bugs with some features. If I can do anything to help with testing, I would be more than happy to help. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Lagerway Sent: 25 July 2003 18:43 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Stuart, X-Lite v2.0 build 1050 was just released. Try that. http://brands.xten.net/x-lite/download/X-Lite_Install.exe Cheers, Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stuart Hirst Sent: Friday, July 25, 2003 10:12 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Dave I tried this and I still have the same problem. I am using X-Lite though and not X-Pro. The SIP registration is fine but still no audio. If anyone has X-Lite either 1016 or 1047 (v2) working, please could you let me know and maybe email your registry settings for the app. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Packham Sent: 25 July 2003 15:38 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Fixed it I have audio now... uninstall everything xten makes and manually clear out all the xten/xlite stuff from the registry.. search for XtenNetwork and kill the keys. reinstall Xpro and it works... go figure Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] reconnecting
Yes, you can start this way and get most of the call flow detail just like when you connect on the main screen. However, if you are writing your own AGI scripts, you wouldn't get any output directed to STDERR (like debugging messages) - these go only to the initial console where * is started. Below are samples of each output. regards, Scott Stingel ON ASTERISK STARTUP SESSION: (INCOMING CALL) *CLI -- Starting simple switch on 'Zap/1-1' NOTICE[1217603008]: File chan_zap.c, Line 4134 (ss_thread): Got event 2 (Ring/Answered)... -- Executing Answer(Zap/1-1, ) in new stack -- Executing Wait(Zap/1-1, 1) in new stack -- Executing AGI(Zap/1-1, test.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.agi *** Got digit string: 1234 THE FOLLOWING 6 LINES ARE DEBUGGING STATEMENTS!: Password in SYSTEM table: 1234 wait_digits exit, # digits = 4 Password OK! stat result size on jt/555= , len= 0 stat result size on ...sounds/jt/555= -LAST DEBUGGING STATEMENT -- AGI Script test.agi completed, returning 0 -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (incoming, s, 4) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' ON SECOND SESSION CONNECTED WITH -rvvvc: *CLI -- Starting simple switch on 'Zap/1-1' -- Executing Answer(Zap/1-1, ) in new stack -- Executing Wait(Zap/1-1, 1) in new stack -- Executing AGI(Zap/1-1, test.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.agi -- AGI Script test.agi completed, returning 0 -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (incoming, s, 4) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' END-- Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darrell Eldridge Sent: Friday, July 25, 2003 9:06 PM To: Asterisk Users Subject: [Asterisk-Users] reconnecting If asterisk is running as a daemon (started with no command line options), can I reconnect to it and get the vvverbose info in the console? In other words, will asterisk -rvvvc work? Sorry to ask I would just try it to see if it works, but I don't have an installation handy and have to do some remote troubleshooting soon. ThanksE+06. D. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T410P and zaptel.conf
It is selected by jumpers on the card. You may override also by using t1e1override=foo when you modprobe wct4xxp Mark On Thu, 24 Jul 2003, Alex Lopez wrote: One a t400p I know that I have 24 channels per port for a total of 96. However the T410 card allows for E1 as well as T1 lines. How does it determine how many channels per port. For a more specific question. Would the first Zap device on the second port be Zap/25 or Zap/30 when using a T1?? I looked for docs on this but found nothing.. Other questions: Is the electrical interface the same for a E1 as a T1?? How does the card know which is which? Is it by the span def in the /etc/zaptel.conf file? Has anyone seen a technical document on this card??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best software SIP client
I am also having the same problem with x-ten. I have tried release 1050 and it still does not work (no audio stream). Steve Besch Erik Lagerway wrote: Stuart, X-Lite v2.0 build 1050 was just released. Try that. http://brands.xten.net/x-lite/download/X-Lite_Install.exe Cheers, Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stuart Hirst Sent: Friday, July 25, 2003 10:12 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Dave I tried this and I still have the same problem. I am using X-Lite though and not X-Pro. The SIP registration is fine but still no audio. If anyone has X-Lite either 1016 or 1047 (v2) working, please could you let me know and maybe email your registry settings for the app. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Packham Sent: 25 July 2003 15:38 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Fixed it I have audio now... uninstall everything xten makes and manually clear out all the xten/xlite stuff from the registry.. search for XtenNetwork and kill the keys. reinstall Xpro and it works... go figure Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Stephen R. Besch, Ph.D. SachsLab Hughes Center for Single Molecule Studies Dept. of Physiology and Biophysics 301F Cary Hall SUNY at Buffalo Buffalo, NY 14214 Phone: (716) 829-3289 x106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Instant hangup on busy Zap channel.
Martin Pycko wrote: Do 'iax2 debug' to see more. Martin Thanks Martin, As it seems as if it may be a bug, I'll get IAX2 debug output from both *'s and put them in the bug tracker to save list clutter. Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can't get musiconhold to work
I can't seem to get musiconhold to work. I'm running asterisk on a RH9 box, I have the mpg123 package installed. In my zapata.conf file I have the line MusicOnHold=default . In my musiconhold.conf file, in the classes section I uncommented default and loud. In my extensions.conf file I have a set musiconhold line. However if I get a call and I either put it on hold or hit flash I get no music. The sample mp3 file is in the mohmp3 directory. Does anyone know what I might be doing wrong or how I might be able to correct it? Also I have tried assigning a extension with the MusicOnHold application and it still doesn't seem to work. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] executing an agi script after a successful Dial
John Thanks for the response. This seems to be what I am looking. However, I have discovered a problem with a simple perl script triggered from the h extension. I am using perl-Asterisk and if I call the script from any extension in works fine. However, if I call the same script from h the get_variable and verbose functions don´t work anymore. Rgds Dan - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 23, 2003 8:20 PM Subject: Re: [Asterisk-Users] executing an agi script after a successful Dial On Wednesday 23 July 2003 05:04 pm, Dan Fernandez wrote: I would like to run an agi script (to calculate the cost of a long distance or international call) right after I execute a Dial app. Can this be configured in extensions.conf? It seems the entries It cannot. If the Dial app succeeds in getting a connected channel, it will ALWAYS return -1, which signals a hangup to Asterisk. The only time Dial will ever return control to the dialplan is if either the channel is not available or if the channel does not get connected. Hmm... I'm not so sure about what the question was, and if perhaps there is some confusion about what is desired here. In my example configs, I use the h extension to clean up call recording after Dial has terminated. Seems to work for me, but perhaps it's not supposed to work. :) Dan - try putting your routines in an extension called h. This may get executed after Dial terminates normally or abnormally. JT right after a Dial app get executed only if the Dial app was executed unsucessfully. Would I have to execute the dial app from the agi script? No, again, the Dial app won't return control to the AGI script until after the call is complete. You're pretty much going to have to do whatever you want to do prior to executing Dial or after the call is complete. Of course, you could create a separate thread which runs parallel to the channel thread and does various monitoring tasks, but that would require some C programming skills. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] executing an agi script after a successful Dial
Hi Dan,
no wonder. when the h extension is called the channel (including all
the channel variables you want to read with get_var) is gone. pass the
channel variables you need to acces as an argument to the agi script,
e.g.: exten = h,1,AGI(myagi.agi,${EXTEN} ${CALLERIDNUM})
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email: [EMAIL PROTECTED]
http://www.junghanns.net/asterisk
Am Sam, 2003-07-26 um 01.28 schrieb Dan Fernandez:
John
Thanks for the response. This seems to be what I am looking. However, I
have discovered a problem with a simple perl script triggered from the h
extension.
I am using perl-Asterisk and if I call the script from any extension in
works fine. However, if I call the same script from h the get_variable and
verbose functions don´t work anymore.
Rgds
Dan
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 23, 2003 8:20 PM
Subject: Re: [Asterisk-Users] executing an agi script after a successful
Dial
On Wednesday 23 July 2003 05:04 pm, Dan Fernandez wrote:
I would like to run an agi script (to calculate the cost of a long
distance or international call) right after I execute a Dial app.
Can this be configured in extensions.conf? It seems the entries
It cannot. If the Dial app succeeds in getting a connected channel,
it will ALWAYS return -1, which signals a hangup to Asterisk. The
only time Dial will ever return control to the dialplan is if either
the channel is not available or if the channel does not get connected.
Hmm... I'm not so sure about what the question was, and if perhaps
there is some confusion about what is desired here. In my example
configs, I use the h extension to clean up call recording after
Dial has terminated. Seems to work for me, but perhaps it's not
supposed to work. :)
Dan - try putting your routines in an extension called h. This may
get executed after Dial terminates normally or abnormally.
JT
right after a Dial app get executed only if the Dial app was
executed unsucessfully. Would I have to execute the dial app from
the agi script?
No, again, the Dial app won't return control to the AGI script until
after the call is complete. You're pretty much going to have to do
whatever you want to do prior to executing Dial or after the call is
complete. Of course, you could create a separate thread which
runs parallel to the channel thread and does various monitoring
tasks, but that would require some C programming skills.
-Tilghman
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Re: [Asterisk-Users] Best software SIP client
I just got build 1050 working I had the same problem until I set Send Internal IP: on in the menu under sip proxy Kyle - Original Message - From: Stuart Hirst [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 25, 2003 1:05 PM Subject: RE: [Asterisk-Users] Best software SIP client Erik, Thanks for the info. I have tried build 1050 on two different PC's and still the same symptoms. One the main machine I have been using for testing I removed the old version, cleaned the registry and installed build 1050 but still no joy. This is a Dell notebook in a docking station. The second PC is also a Dell Notebook but not as high spec and is not in a docking station. This had not had X-Lite installed before and so I just installed build 1050 with exactly the same results. I should say that build 1005 is still working fine apart from the odd bugs with some features. If I can do anything to help with testing, I would be more than happy to help. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Lagerway Sent: 25 July 2003 18:43 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Stuart, X-Lite v2.0 build 1050 was just released. Try that. http://brands.xten.net/x-lite/download/X-Lite_Install.exe Cheers, Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stuart Hirst Sent: Friday, July 25, 2003 10:12 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Dave I tried this and I still have the same problem. I am using X-Lite though and not X-Pro. The SIP registration is fine but still no audio. If anyone has X-Lite either 1016 or 1047 (v2) working, please could you let me know and maybe email your registry settings for the app. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Packham Sent: 25 July 2003 15:38 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Best software SIP client Fixed it I have audio now... uninstall everything xten makes and manually clear out all the xten/xlite stuff from the registry.. search for XtenNetwork and kill the keys. reinstall Xpro and it works... go figure Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audiocodes fxs
Kelvin Chua wrote: hi guys, have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing? ~kelvin Yes, Ok. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best software SIP client
On Fri, 25 Jul 2003, Dave Packham wrote: Fixed it I have audio now... uninstall everything xten makes and manually clear out all the xten/xlite stuff from the registry.. search for XtenNetwork and kill the keys. reinstall Xpro and it works... go figure For what it's worth, I was having lots of problems until I did just this... but I think it's because I screwed up configuration of X-Lite. But an uninstall/registry clean followed by a reinstall and a proper configuration of all of the relevant settings worked. I'm now doing inbound and outbound calls through our Asterisk using X-Lite (the most recent build, which I believe is 1050). -- JustThe.net Internet Multimedia Svcs. [The Fusion of Content Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] executing an agi script after a successful Dial
Thanks for the response. In addition to what you stated, I think there is
another problem with Asterisk::AGI
This is the test script
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $num = $AGI-get_variable('FOO')
$AGI-verbose(get_variable\FOO\=$num,1);
--
extensions.conf
exten= h, 1,SetVar(FOO=)
exten= h,2,Agi,test.agi
exten = _6XX,1,Agi,db.agi
exten = _4XX,1,Dial,${TEST}
--
If I call the Agi by dialing 666 the perl script works just fine and it runs
twice (I think this is strange since I didn´t execute a Dial)
If I dial 444 the script executes but I get no output.
Therefore it seems there is a problem with Asterisk::AGI
- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 25, 2003 8:32 PM
Subject: Re: [Asterisk-Users] executing an agi script after a successful
Dial
Hi Dan,
no wonder. when the h extension is called the channel (including all
the channel variables you want to read with get_var) is gone. pass the
channel variables you need to acces as an argument to the agi script,
e.g.: exten = h,1,AGI(myagi.agi,${EXTEN} ${CALLERIDNUM})
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon: +49 30 79705392
fax: +49 30 79705391
iaxtel: 1-700-157-8753
email: [EMAIL PROTECTED]
http://www.junghanns.net/asterisk
Am Sam, 2003-07-26 um 01.28 schrieb Dan Fernandez:
John
Thanks for the response. This seems to be what I am looking. However, I
have discovered a problem with a simple perl script triggered from the h
extension.
I am using perl-Asterisk and if I call the script from any extension in
works fine. However, if I call the same script from h the get_variable and
verbose functions don´t work anymore.
Rgds
Dan
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 23, 2003 8:20 PM
Subject: Re: [Asterisk-Users] executing an agi script after a successful
Dial
On Wednesday 23 July 2003 05:04 pm, Dan Fernandez wrote:
I would like to run an agi script (to calculate the cost of a long
distance or international call) right after I execute a Dial app.
Can this be configured in extensions.conf? It seems the entries
It cannot. If the Dial app succeeds in getting a connected channel,
it will ALWAYS return -1, which signals a hangup to Asterisk. The
only time Dial will ever return control to the dialplan is if either
the channel is not available or if the channel does not get connected.
Hmm... I'm not so sure about what the question was, and if perhaps
there is some confusion about what is desired here. In my example
configs, I use the h extension to clean up call recording after
Dial has terminated. Seems to work for me, but perhaps it's not
supposed to work. :)
Dan - try putting your routines in an extension called h. This may
get executed after Dial terminates normally or abnormally.
JT
right after a Dial app get executed only if the Dial app was
executed unsucessfully. Would I have to execute the dial app from
the agi script?
No, again, the Dial app won't return control to the AGI script until
after the call is complete. You're pretty much going to have to do
whatever you want to do prior to executing Dial or after the call is
complete. Of course, you could create a separate thread which
runs parallel to the channel thread and does various monitoring
tasks, but that would require some C programming skills.
-Tilghman
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[Asterisk-Users] Busy detect on pri channel?
Did anybody figure out how to make dial detect a busy on a zaptel channel on a pri interface when using overlap dialing? According to the documentation dial should return to priority n+101, if the called party is found to be busy. I can see a DISCONNECT message with user busy coming from the network when I turn on pri debugging, but the dial application does not seem to notice. On a related issue: Is there a way of retriving isdn causes for the last call? I think it would be very useful for a lot of applications to distinguish in between say network congested, number unallocated and a regular hangup. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco's CallManager and *
On Wed, Jul 23, 2003 at 10:56:47PM -0400, Jeremy McNamara wrote: Either don't use unity and point all the IP phones to * for VM or just setup an H.323 trunk that dumps u into the appropriate mailbox, if u must use it. This is what I'd like to see happen. I'm slowly but surely getting familar with asterisk, and hopefully we can put something together that's better than CCM and Unity :) Btw, your company wasted good money on that crap. (IMHO)Just hope Yeah, I think so, I did try to mention asterisk when we were searching for a new phone system :( they don't ever plan on acutally billing anyone for services with CCM. We're the users, so there's nobody to bill to :( -- Yifang Dai | eFax: (847)628-0255 |Debian GNU/Linux [EMAIL PROTECTED] | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] executing an agi script after asuccessful Dial
is there any way to keep those vars around until after h goes away?maybe move the
free routiene to after h is done?
Dave
[EMAIL PROTECTED] 7/25/2003 5:32:55 PM
Hi Dan,
no wonder. when the h extension is called the channel (including all
the channel variables you want to read with get_var) is gone. pass the
channel variables you need to acces as an argument to the agi script,
e.g.: exten = h,1,AGI(myagi.agi,${EXTEN} ${CALLERIDNUM})
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email: [EMAIL PROTECTED]
http://www.junghanns.net/asterisk
Am Sam, 2003-07-26 um 01.28 schrieb Dan Fernandez:
John
Thanks for the response. This seems to be what I am looking. However, I
have discovered a problem with a simple perl script triggered from the h
extension.
I am using perl-Asterisk and if I call the script from any extension in
works fine. However, if I call the same script from h the get_variable and
verbose functions don t work anymore.
Rgds
Dan
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 23, 2003 8:20 PM
Subject: Re: [Asterisk-Users] executing an agi script after a successful
Dial
On Wednesday 23 July 2003 05:04 pm, Dan Fernandez wrote:
I would like to run an agi script (to calculate the cost of a long
distance or international call) right after I execute a Dial app.
Can this be configured in extensions.conf? It seems the entries
It cannot. If the Dial app succeeds in getting a connected channel,
it will ALWAYS return -1, which signals a hangup to Asterisk. The
only time Dial will ever return control to the dialplan is if either
the channel is not available or if the channel does not get connected.
Hmm... I'm not so sure about what the question was, and if perhaps
there is some confusion about what is desired here. In my example
configs, I use the h extension to clean up call recording after
Dial has terminated. Seems to work for me, but perhaps it's not
supposed to work. :)
Dan - try putting your routines in an extension called h. This may
get executed after Dial terminates normally or abnormally.
JT
right after a Dial app get executed only if the Dial app was
executed unsucessfully. Would I have to execute the dial app from
the agi script?
No, again, the Dial app won't return control to the AGI script until
after the call is complete. You're pretty much going to have to do
whatever you want to do prior to executing Dial or after the call is
complete. Of course, you could create a separate thread which
runs parallel to the channel thread and does various monitoring
tasks, but that would require some C programming skills.
-Tilghman
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Re: [Asterisk-Users] time and date stamp in voicemail
On Friday 25 July 2003 14:12, Andy Hester wrote: Dan, the page is actually http://asterisk.drunkcoder.com/patches/ . However, I didn't see the patch there. I just added it. It's available there now. Note that there are three files: a patch, sounds, and some instructions. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time and date stamp in voicemail
Tilghman, Thanks alot for posting that. I'll check it out Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher Sent: Friday, July 25, 2003 10:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] time and date stamp in voicemail On Friday 25 July 2003 14:12, Andy Hester wrote: Dan, the page is actually http://asterisk.drunkcoder.com/patches/ . However, I didn't see the patch there. I just added it. It's available there now. Note that there are three files: a patch, sounds, and some instructions. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] time and date stamp in voicemail
Tilghman Lesher wrote: On Friday 25 July 2003 14:12, Andy Hester wrote: Dan, the page is actually http://asterisk.drunkcoder.com/patches/ . However, I didn't see the patch there. I just added it. It's available there now. Note that there are three files: a patch, sounds, and some instructions. I wonder if you could comment a bit as to the fact that the patch is set against the older voicemail apps instead of the 2 series. I want to use the many-directory voicemail method, but I sure would like to use the timestamps too. Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
