[Asterisk-Users] Newbie - Voicemail Password Help
Hello All.
I'm just beginning with Asterisk and I have it all working now. I'm using
Asterisk 1.0 RC1.
My only question is this; when I check my voice mail the PBX simply says
password. I wanted to make it say please enter your voice mail password so
I am using Background(pls-enter-vm-password).
However now I hear Please enter your voice mail password password when I
check my messages.
That's not a type-o. It says password twice.
Here is my extensions.conf file.
[macro-vmanswer]
exten = s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
exten = s,2,Background(pls-enter-vm-password)
exten = s,3,VoicemailMain(${ARG1})
exten = s,4,Hangup
exten = s,5,Voicemail(u${ARG1})
exten = s,6,Hangup
[default]
exten = 1002,1,Macro(vmanswer,1002)
The whole point of the vmanswer macro is to go to the voice mail main menu
automatically when calling from your own phone, otherwise it sends callers to
the voice mail system to leave a message. Perhaps there's a better way to do
this as well. If so, please let me know.
Regards,
Paul
__
Do you Yahoo!?
Yahoo! Mail Address AutoComplete - You start. We finish.
http://promotions.yahoo.com/new_mail
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How record conversation to sound file ?
lst.ara [EMAIL PROTECTED] writes: For our helpdesk application, we need record full conversation between any caller and one or two helpdek numbers (while the conversation is running). After conversation is ended (hangup ..), the recorded file (WAW) is putted into database. Using AGI, record and put to database is OK, but only as exclusive task. But I need record WAW file in background the standard conversation. Is any vay to do it with Asterisk ? Hi, did you try the Asterisk commands Record or Monitor? http://www.voip-info.org/wiki-Asterisk+cmd+Record http://www.voip-info.org/wiki-Asterisk+cmd+Monitor I am not sure if they do what you would like to have but I thought it's better to give you the hint. I have no experience with these commands. Regards, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] SMS Asterisk
Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Maxim Litnitsky Gesendet: Sonntag, 29. August 2004 23:59 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] SMS Asterisk Hi all! I am intrested in the following scheme My mobile phone - SMS to SOMETHING - Redirect to FWD number - FDW redirect to my * - My * doing smtg Why not send the sms to * directly? It works in .de and .uk for sure. Not sure whether it works in other countries too, haven't tried it yet. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transferring call to another line
Hi, I just got my zaptel fxo cards working, and I want to be able to have someone call in on one line and access the other - I guess what I want to do is transfer(exten), but that is only for extensions - not channels which is what I want i guess. I tried the Dial(Zap/2) but I think that's for ringing that line (fxs)? thanks Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 SIP
I too have been playing with the cisco 7940's. I am guessing, as they only have two line appearence keys they are only able to handle two lines. However they appear to be able to handle 4 simultaneous calls (two per line key). If you assign two different numbers, one to each key then you can only have two simultaneous calls per number. I am running firmware V7.02 I guess that cisco figure if you need to handle more calls/lines you will either buy their 7960 or 7914 add on module. Has anyone figured out how to 'hide' the second line appearence. I wish to setup an intercom line but have to assign it a slightly different no. I dont want this line to be selectable from the phone, Sam Brian Pavane [EMAIL PROTECTED] wrote on 27/08/2004 20:20:15: Chris, I have reached the same situation that you have, and not been able to have more than 2 inbound calls into a Cisco SIP phone. It also appears that if you are on Line 1 with a call, and an inbound call comes in, it goes to Line 2 instead of being the second call on Line 1. -Brian Christopher L. Wade wrote: Hi again, I know I asked a similar question earlier this week/last week. But in that email, I forgot to mention, even though I'm sure it was assumed, that I'm using the SIP image on the phones. My question, as stated last time, is just how many *incoming* calls can the 7940 have? I've had 4 outgoing calls at once, I've had 2 incoming and 2 outgoing. But no matter what I do, I cannot receiving more than 2 incoming calls. Is this the limit of the 7940? My configuration, right now, is to have both line buttons assigned to the same extension, thus allowing one incoming and one outgoing per line button. Thanks for the help, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Winckworth Sherwood Solicitors and Parliamentary Agents DX 148400 WESTMINSTER 5 : 35 Great Peter Street, London SW1P 3LR Telephone 020 7593 5000 Fax 020 7593 5099 Confidentiality This email message and any attachments are confidential; they may be subject to legal professional privilege and are intended for the named recipient only. If you are not the named recipient, please return the message and enclosures immediately and delete them from your system. Caution Before advice received only by email (whether by attachment or otherwise) may be relied on, the authenticity of the communication must be verified by means independent of email. Regulation The firm is regulated by the Law Society. Partners A list of partners is available for inspection at each office of the firm and on the firm's website at www.winckworths.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How record conversation to sound file ?
Martin Holler napsal(a): lst.ara [EMAIL PROTECTED] writes: For our helpdesk application, we need record full conversation between any caller and one or two helpdek numbers (while the conversation is running). After conversation is ended (hangup ..), the recorded file (WAW) is putted into database. Using AGI, record and put to database is OK, but only as exclusive task. But I need record WAW file in background the standard conversation. Is any vay to do it with Asterisk ? Hi, did you try the Asterisk commands Record or Monitor? http://www.voip-info.org/wiki-Asterisk+cmd+Record http://www.voip-info.org/wiki-Asterisk+cmd+Monitor Thank you. The Monitor command is what I need. It is new feature and I Did not know it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Do not get calldeflection (capiCD) to work.
I do not get calldeflection (capiCD) to work. The Mobile do not ring, it seems the CD do not work. I use the chan_capi 0.3.5 and have no idea. please help me. nico ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] SMS Asterisk
[...] Why not send the sms to * directly? It works in .de and .uk for sure. [...] Can you enlighten us as to how exactly? Axel -- Axel Eble, CISSP * Trienter Str. 6b * 87437 Kempten (Allgäu) * Germany VoIP: [EMAIL PROTECTED] * cell: +49.178.285-3265 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Jitter over Sat
Hello, I have a problem with jitter over a 2mb up 1mb down satellite connection. I call my friend over the satellite - I call perfect but they cannot make out a word I say. However if I leave him voicemail on his asterisk box, it records my voice perfect. I have this problem when calling other people as well. This is my setup: [my Grandstream]- [my * PBX]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) I've also tried: [my Grandstream]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) and: [my Grandstream]- [my * PBX]- [sat]- [friends Supra Phone] (or any other device) I've tried all combination of using SIP and IAX2 connections to bridge the calls using codecs ULAW and iLBC with all the same result. When I call my friends ECHO BACK TEST, I sound perfect (with a bit of latency). Anyone have some suggestions? Thanks kindly, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer to queue
Hi, Using a cisco 7960, if I try and transfer someone using the transfer button, when I transfer them to a queue, it seems to disconnect them. Does anyone know why? I simply have an extension that points to a queue (e.i. exten = 281,1,Queue(Sales) ). Cheers, Ben Merrills ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint... Gains - Which isfor speakerphone
Thanks, I was afraid to try and change gains that I didn't know what they did simply because I don't want to blow a speaker or something.. :) I'll try it today. The only thing I haven't figured out is how to set a default ringer in the configuration file, set my time to EST w/ Daylight Savings and when receiving incoming calls if it's possible to see NAME NUMBER instead of just NAME. I'd prefer to see just NUMBER over name. All incoming calls are masked with 'Toll-Free Call' for some reason. So every caller, I get Toll-Free Call. And for my tech support options, etc... I change the name to 'Tech Support' to easily be able to tell what department a person called. But now I can't see the phone number, only AFTER I pick up. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Monday, August 30, 2004 11:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom SoundPoint... Gains - Which isfor speakerphone Hmmm... Hands Free might be: voice.gain.rx.digital.chassis=15 (15 is my setting) Call waiting? You can turn it off in sip.cfg - do not disturb settings I think. Don't know about gain for call waiting. You might try playing with some of the variables in ipmid.cfg under ringType John Matthew Marlowe wrote: now that I have finally figured out what I was doing wrong with my polycom phone and got it to read the configuration file Im changing some gains. I successfully changed the gain for the ringer... It was too low for me. Does anyone know which gain would be for the call waiting and which tone would be for the hands free mode? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ** ASTRICON * LAST CALL FOR REGISTRATION
Astricon is the first Asterisk user's and developer's conference to be held in Atlanda, GA, USA Sept 22-24 this fall. Astricon is organized by Edvina.net and Sokol Associates in partnership with Digium, inc. ** CURRENT ESTIMATE: OVER 250 ATTENDEES Steven and I started this project at the end of march, looking forward to gathering the community and showing the strength of an Open Source project from a business and a community standpoint. In our wildest dreams, we couldn't really imagine the success we've had so far. We've almost sold out the sponsorships. We've almost sold out the seats for attendees! We'll be well over 250 Asterisk users in the conference! ** NEW TUTORIALS The tutorial agenda have changed. The internatiolatization tutorial is replaced by a tutorial covering Asterisk GUI's - user interfaces for administrators, users and receptionists. This is a multi-speaker session moderated by Jim Thompson, the voip-info.org Wiki maintainer. (There's still room for one more GUI project. Mail me if you are interested!) ** RUMOURS, LAUNCHES, GIVE-AWAYS There will be many new products launched at Astricon. From behind the curtains, we've seen and heard a lot of new stuff in the works. Make sure that you visit the Astricon Exhibition! There's also rumours about a launch party, sponsored by the major Asterisk.org contributor. Be there! ** REGISTER NOW, WE ARE REACHING THE UPPER LIMIT We are now fast reaching the upper limit in number of attendees. Make sure you register now to get a hotel room and an entrance ticket. http://www.astricon.net Thank you for all your support! Steven Sokol Olle E. Johansson Astricon Organizers PS: And if we haven't been as active on the IRC or the bug tracker as usual, you now understand why... :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Jitter over Sat
Opps, at 3am I make stupid editing mistakes. Should read: I have a problem with jitter over a 2mb up 1mb down satellite connection. I call my friend over the satellite - **I can hear him perfect**, but he cannot make out a word I say. However if I leave him voicemail on his asterisk box, it records my voice perfect. I have this problem when calling other people as well. Storm D. J. Petersen mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Storm D. J. Petersen Sent: Tuesday, August 31, 2004 3:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Jitter over Sat Hello, I have a problem with jitter over a 2mb up 1mb down satellite connection. I call my friend over the satellite - I call perfect but they cannot make out a word I say. However if I leave him voicemail on his asterisk box, it records my voice perfect. I have this problem when calling other people as well. This is my setup: [my Grandstream]- [my * PBX]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) I've also tried: [my Grandstream]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) and: [my Grandstream]- [my * PBX]- [sat]- [friends Supra Phone] (or any other device) I've tried all combination of using SIP and IAX2 connections to bridge the calls using codecs ULAW and iLBC with all the same result. When I call my friends ECHO BACK TEST, I sound perfect (with a bit of latency). Anyone have some suggestions? Thanks kindly, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer from MOH to MOH doesn't work.
Hi, If I try to transfer a user (user listens to MOH while I transfer) to eg. a queue, and the transfer occour while the MOH in the queue is playing, the MOH will stop, and the user hears nothing but scilence, but is in the queue. If I make the transfer to the queue, while still listening to the announcement, the user will hear the announcement, and then the MOH will start. Using latest CVS Incomming call thorugh Fritz Card Called phone: Cisco 7940 -- Med venlig hilsen / Best regards Michael Løjtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 Søborg Tel (+45) 3955 0700 - Fax (+45) 3955 0707 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Jitter over Sat
Hi there, this is just a me too... well, not exactly. I get jitter when trying to make SIP calls through Asterisk using a GPRS connection... can this be done actually? TIA, Martin Storm D. J. Petersen wrote: Hello, I have a problem with jitter over a 2mb up 1mb down satellite connection. I call my friend over the satellite - I call perfect but they cannot make out a word I say. However if I leave him voicemail on his asterisk box, it records my voice perfect. I have this problem when calling other people as well. This is my setup: [my Grandstream]- [my * PBX]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) I've also tried: [my Grandstream]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) and: [my Grandstream]- [my * PBX]- [sat]- [friends Supra Phone] (or any other device) I've tried all combination of using SIP and IAX2 connections to bridge the calls using codecs ULAW and iLBC with all the same result. When I call my friends ECHO BACK TEST, I sound perfect (with a bit of latency). Anyone have some suggestions? Thanks kindly, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile error H323
Enrico Stahn wrote: Hi! Have a look at the following entry. I solved this problem: http://enrico.todo.de/weblog/item/asterisk-oh323-compile-error That's the wrong way to do it. You use incorrect versions of the libraries. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pattern matching problems
this is from my extensions.conf, the first three patterns are for
toll-free numbers, and fourth pattern is for other numbers, where an AGI
is called for authentication.
now when I dial 011448000664327 if falls into the fourth pattern, where
as it should be matched by the first pattern. Any suggestions
1 - exten = _01144800XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
2 - exten = _01144808XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
3 - exten = _01144500XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
4 - exten = _011.,1,AGI(iax.agi)
4 - exten = _011.,2,Dial(${MAG}/${EXTEN:3},45,tT)
4 - exten = _011.,103,playback(no-service)
thank you
--
Atif
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pattern matching problems
Hi,
-Original Message-
this is from my extensions.conf, the first three patterns are for
toll-free numbers, and fourth pattern is for other numbers,
where an AGI
is called for authentication.
now when I dial 011448000664327 if falls into the fourth
pattern, where
as it should be matched by the first pattern. Any suggestions
1 - exten = _01144800XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
2 - exten = _01144808XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
3 - exten = _01144500XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
4 - exten = _011.,1,AGI(iax.agi)
4 - exten = _011.,2,Dial(${MAG}/${EXTEN:3},45,tT)
4 - exten = _011.,103,playback(no-service)
Better to do it like this:
[mycontext]
Include = numberedcases
Include = othercases
[numberedcases]
exten = _01144800XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _01144808XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _01144500XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
[othercases]
exten = _011.,1,AGI(iax.agi)
exten = _011.,2,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _011.,103,playback(no-service)
Included contexts are matched sequentially.
Florian
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pattern matching problems
On Tue, 2004-08-31 at 21:42, Atif Rasheed wrote:
this is from my extensions.conf, the first three patterns are for
toll-free numbers, and fourth pattern is for other numbers, where an AGI
is called for authentication.
now when I dial 011448000664327 if falls into the fourth pattern, where
as it should be matched by the first pattern. Any suggestions
1 - exten = _01144800XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
2 - exten = _01144808XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
3 - exten = _01144500XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
4 - exten = _011.,1,AGI(iax.agi)
4 - exten = _011.,2,Dial(${MAG}/${EXTEN:3},45,tT)
4 - exten = _011.,103,playback(no-service)
This is because asterisk is 'lazy'. It will not take the first
matching extension, nor will it take the most specific matching
extension, instead, it will take the least specific extension. This
means, regardless of the number, if it matches 011* then it will always
take that option. The only way to acheive what you want (AFAIK) is like
this:
[blah]
include = foo
include = bar
[foo]
exten = _01144800XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _01144808XXX,1,Dial(${MAG/${EXTEN:3},45,tT)
exten = _01144500XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
[bar]
exten = _011.,1,AGI(iax.agi)
exten = _011.,2,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _011.,103,playback(no-service)
This will force asterisk to look/match extensions in foo before it
attempts to look/match extensions in bar.
Hope this helps (and it is actually correct, try it and see)
Regards,
Adam
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie - Voicemail Password Help
You did not replace the existing prompt, but added a second prompt. The
proper place to make this adjustment is in VoicemailMain, not in extensions.
Or find the password prompt sound file and just replace it with yours.
Lyle
- Original Message -
From: Java Rockx [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 1:10 AM
Subject: [Asterisk-Users] Newbie - Voicemail Password Help
Hello All.
I'm just beginning with Asterisk and I have it all working now. I'm using
Asterisk 1.0 RC1.
My only question is this; when I check my voice mail the PBX simply says
password. I wanted to make it say please enter your voice mail
password so
I am using Background(pls-enter-vm-password).
However now I hear Please enter your voice mail password password when I
check my messages.
That's not a type-o. It says password twice.
Here is my extensions.conf file.
[macro-vmanswer]
exten = s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
exten = s,2,Background(pls-enter-vm-password)
exten = s,3,VoicemailMain(${ARG1})
exten = s,4,Hangup
exten = s,5,Voicemail(u${ARG1})
exten = s,6,Hangup
[default]
exten = 1002,1,Macro(vmanswer,1002)
The whole point of the vmanswer macro is to go to the voice mail main menu
automatically when calling from your own phone, otherwise it sends callers
to
the voice mail system to leave a message. Perhaps there's a better way to
do
this as well. If so, please let me know.
Regards,
Paul
__
Do you Yahoo!?
Yahoo! Mail Address AutoComplete - You start. We finish.
http://promotions.yahoo.com/new_mail
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Jitter over Sat
On Tuesday 31 August 2004 11:36, Martin Mielke wrote: Hi there, this is just a me too... well, not exactly. I get jitter when trying to make SIP calls through Asterisk using a GPRS connection... can this be done actually? [...] Yes, we've done it over Vodaphone (I think). The lag, about 1.5s in some tests weve done, can really kill it. B ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions = s,1,Dial(Zap/2/number) noise
Hi, I'm trying to answer a call on one line and dial out a number on a zaptel x100p fxo, but all I get from the phone I'm dialing is silence after it is picked up, and on the line that's supposed to be dialed out itself, noise. Thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - Voicemail Password Help
Paul,
What you can do is modify the source code for the voicemail application.
Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the file
'vm-password' to 'pls-enter-vm-password'.
Recompile and install.
Then in your macro remove the line that plays the 'pls-enter-vm-password' file.
Steve
From: [EMAIL PROTECTED] on behalf of Java Rockx
Sent: Mon 8/30/2004 8:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Newbie - Voicemail Password Help
Hello All.
I'm just beginning with Asterisk and I have it all working now. I'm using
Asterisk 1.0 RC1.
My only question is this; when I check my voice mail the PBX simply says
password. I wanted to make it say please enter your voice mail password so
I am using Background(pls-enter-vm-password).
However now I hear Please enter your voice mail password password when I
check my messages.
That's not a type-o. It says password twice.
Here is my extensions.conf file.
[macro-vmanswer]
exten = s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
exten = s,2,Background(pls-enter-vm-password)
exten = s,3,VoicemailMain(${ARG1})
exten = s,4,Hangup
exten = s,5,Voicemail(u${ARG1})
exten = s,6,Hangup
[default]
exten = 1002,1,Macro(vmanswer,1002)
The whole point of the vmanswer macro is to go to the voice mail main menu
automatically when calling from your own phone, otherwise it sends callers to
the voice mail system to leave a message. Perhaps there's a better way to do
this as well. If so, please let me know.
Regards,
Paul
__
Do you Yahoo!?
Yahoo! Mail Address AutoComplete - You start. We finish.
http://promotions.yahoo.com/new_mail
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
winmail.dat___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Jitter over Sat
I don't mind latency ... it's the garbage jitter where no one can understand a word. Interestingly enough if I do this it works fine: [grandstream 1]- [sat]- [pbx in mothers house] [grandstream 2]- [sat] -/ where the grandstream phones are side by side. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bob Goddard Sent: Tuesday, August 31, 2004 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Jitter over Sat On Tuesday 31 August 2004 11:36, Martin Mielke wrote: Hi there, this is just a me too... well, not exactly. I get jitter when trying to make SIP calls through Asterisk using a GPRS connection... can this be done actually? [...] Yes, we've done it over Vodaphone (I think). The lag, about 1.5s in some tests weve done, can really kill it. B ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
Chris Shaw wrote: - Channel Support: IAX2 in asterisk IAX2 in libiax2 Other IP channels in asterisk (RTP-based ones, I guess are all that is left). CNG/VAD and DTX in SIP is a must if * is to be taken seriously as a complete solution... As much as we all hate it's complexity and wish that everything would speak IAX (I know I do) a large number of devices support (and will be supporting) SIP, making it equally as important as IAX2 in using * as a complete telephony solution... This is nothing to do with SIP. It is an RTP issue, common to everything which uses RTP - SIP and H.323 included. Sending no packets is perfectly valid, and normal, in RTP. If the receiving end takes no packets (other than, perhaps, an extremely long silence) as a disconnect it does not comply with the RTP spec. DTX is much despised, and CNG only slightly better. They just sound good (pun intende) on paper. DTX Support: Sending a single CN packet (in IAX2, this should probably sent reliably) would probably be good. I second, third and fourth this one as does anyone who's tried to use BroadVoice with Voicemail... Currently when * is not making any noise (e.g. recording) absolutely NO packets are sent back to the proxy... A lot of proxies take this as a sign that the far end has disconnected... Including BroadWorks! But they do recognize small CN packets as a sign that the SIP device (Asterisk) is still there... A lot of CNG spec. call for only one transmission, and then silence. Continued CNG has real benefits, but it certainly not the norm. PLC I think is somewhat implemented already in codecs that support it, but I could be wrong, I remember seeing mention of it in the code... PLC is seldom included in the codecs. If you read the specs they often mention PLC, but only in terms of how the codec mitigates the awfulness of a lost packet. Few codecs actually include it. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: pattern matching problems
thank you people for your help, I have done it, and in a different way,
like
exten = _01144800XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _01144808XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _01144500XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _011X.,1,AGI(iax.agi)
exten = _011X.,2,Dial(${MAG}/${EXTEN:3},45,tT)
exten = _011X.,103,playback(no-service)
I made the _011. more precise, I should say
--
Atif
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BRI numbers
Good day all I'm new to the whole pbx thing.I've setup 2 servers with voicetronix card! Each card's got 4 ports.Ive configured it so each port is for a different company,so in other words if a call comes in on port 1 it plays company 1's welcome message ens..I did this with context in vpb.conf Now I'm looking into ISDN bri. Please correct me if I'm wrong. The BRI ISDN card I'm looking at has 2 line-ports.Now I'm not sure of the amount but each line can have about 20 numbers. Now my question is how do I do the same type of config for BRI cards as for the vpicetronix cards In other word,1 line,5 numbers,5 company's,5 different welcome messages Please Let me know Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] SMS Asterisk
Pick up mobile phone.. enter sms .. send it to the * phone number. Done On the * side.. follow the sms howto (voip-info.org might have some infos) Done -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Axel Eble Gesendet: Dienstag, 31. August 2004 11:27 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: AW: [Asterisk-Users] SMS Asterisk [...] Why not send the sms to * directly? It works in .de and .uk for sure. [...] Can you enlighten us as to how exactly? Axel -- Axel Eble, CISSP * Trienter Str. 6b * 87437 Kempten (Allgäu) * Germany VoIP: [EMAIL PROTECTED] * cell: +49.178.285-3265 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: AW: [Asterisk-Users] SMS Asterisk
On Tue, 31 Aug 2004 15:22:26 +0200, Michael Labuschke [EMAIL PROTECTED] wrote: Pick up mobile phone.. enter sms .. send it to the * phone number. Done On the * side.. follow the sms howto (voip-info.org might have some infos) Done Ah. That requires SMS to be available on land lines. Axel -- Axel Eble, CISSP * Trienter Str. 6b * 87437 Kempten (Allgäu) * Germany VoIP: [EMAIL PROTECTED] * cell: +49.178.285-3265 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: AW: [Asterisk-Users] SMS Asterisk
Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Axel Eble Gesendet: Dienstag, 31. August 2004 15:35 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: AW: AW: [Asterisk-Users] SMS Asterisk On Tue, 31 Aug 2004 15:22:26 +0200, Michael Labuschke [EMAIL PROTECTED] wrote: Pick up mobile phone.. enter sms .. send it to the * phone number. Done On the * side.. follow the sms howto (voip-info.org might have some infos) Done Ah. That requires SMS to be available on land lines. Axel Which is.. in .de and .uk Michael -- Axel Eble, CISSP * Trienter Str. 6b * 87437 Kempten (Allgäu) * Germany VoIP: [EMAIL PROTECTED] * cell: +49.178.285-3265 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP registration with public dynamic ip address
Hi, I'm trying to configure a natted budgetone phone to a asterisk server as described in wiki using port forwarding. I successfully make call from the client but it seems it does not register the client ip address and when I try to recall it is not reacheable. Asterisk can manage natted sip client with dynamic ip address ? Bye ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] limit the length of extensions
How do I limit the length of an extension? In my test IVR/Automated
Attendant (whatever it's called), at the beginning it plays if you know
your parties 3 digit extension, you may enter it now) and then it gives
a list of options. If the caller puts the 3 digit extension, it goes
through fine, if they press 1, or 2 it goes to the selected menu option,
but if they dial 91235551212 it dials that phone number. Which of
course, is a big security risk.
Is there a way to limit the length of an extension for an incoming call?
My only solution right now is to duplicate ever single extension (about
50 of them) in a seperate context, one that does not have the _9.
extension in it, and then make the call in menu have access to that
context. However, if I put a limit in the entire context of 3 digits,
then my coworkers who's phones are in that context can only dial each
other, not 9 and an outside number. So it has to be an incoming limit or
something.
Another possibly creative solution would be to SetGroup(outsidecaller)
on the incoming line and then just before my outbound extension put
SetGroup(outsidecaller) and then a CheckGroup(2) or something like
that. I'd have to put another SetGroup in the outbound extension
because there's no way to specify the group with the checkgroup command,
it gets it from the setgroup statement.
Any help would be appreciated.
Thanks,
Deon
[incoming]
exten = 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
exten = 9543340726,2,setcidname(Blocked)
exten = 9543340726,3,setcidnum(00)
exten = 9543340726,4,Goto(companyname,beginmenu,1)
[companyname] ; All the phones, including outbound extensions are in
this context
exten = beginmenu,1,SetVar(CALLEDNAME=CompanyName)
exten = beginmenu,2,Wait,1
exten = beginmenu,3,Background(company-main)
exten = beginmenu,4,Background(ifyouknow)
exten = beginmenu,5,Goto(company_mainmenu,s,1)
exten =
_9NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
exten = 502,1,Dial(SIP/whatever1SIP/whatever2|30|m)
...
[company_mainmenu]
exten = s,1,Background(company-nav1)
exten = 1,1,Goto(company_sales,s,1) ; Sales
exten = 2,1,Goto(companyname,502,1) ; Accounting
exten = 3,1,Goto(companyname,508,1) ; Customer Care
exten = 4,1,Goto(companyname,507,1) ; Technical Support
exten = 5,1,Goto(companyname,202,1) ; Human Resources
exten = 6,1,Goto(companyname,202,1) ; Provisioning
exten = 7,1,Goto(companyname,214,1) ; Marketing
exten = 0,1,Goto(companyname,210,1) ; Operator
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] which distro for asterisk?
Hi I want to play a bit with Asterisk. I currentlly install a new system for that and I would like to get your recommendations regarding the linux distro to use there. This is NOT intended to become a general distro flame war. My favorite distro is and no argument that you flame will convince me here (probably because I've heard it before). However I would like to minimize the OS maintinance task. I really wouldn't like to start worrying about upgrading sshd due to some stupid secuirty hole, and to worry what will it break on my system. I expect my distro to do that for me. I'd also like to have solid astrisk packages that won't break unnecessarily when the sshd package is updated next time. Hopefully also some sort of integration of zaptel in the distro's kernel package. I saw numerous complaints about unofficial RPM packages of asterisk. Besides them, the following free distros include asterisk packages: 1. Debian: http://packages.debian.org/asterisk . 2. Gentoo: Current package seems to be version 0.9.0 from 10-May-2004 3. The DAG repository for RH/Fedora: http://dag.wieers.com/packages/asterisk/ I have some experince with Debian, Mandrake and RedHat/Fedora. I'm unfamiliar with Gentoo and I have no good/bad experince with DAG packages with respect to quality and stability. Any recommendations, relevant experince and other learned opinions? thx -- Tzafrir Cohen +---+ http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend| mailto:[EMAIL PROTECTED] +---+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse Connect DTMF with IAX2
Here's my iax.conf and extensions.conf (I have not yet made the recent changes they just emailed about a day ago, this is twice in a two month period, jeesh) I have tested inbound and outbound dtmf. I use the g.711 codec and use inband. iax.conf -- [general] port=5036 bindaddr=0.0.0.0 context=incoming ;iaxcompat=yes ; Set iaxcompat to yes if you plan to use layered switches. ; It incurs a small performance hit to enable it. delayreject=yes ; For increased security against brute force password attacks. ; Enabling this will delay the sending of authentication ; reject for REGREQ or AUTHREP if there is a password. amaflags=documentation ; global default AMA flag for iaxtel calls. These flags ; are used in the generation of call detail records. ;accountcode=1 ; default account for Call Detail Records in addition ; to specifying on a per-user basis. language=en ; Global default language for users. ; If omitted, will fallback to english bandwidth=high ; Specify bandwidth of low, medium, or high to ; control which codecs are used in general. allow=all ; Which codecs to allow, same as bandwidth=high disallow=g723.1 ; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. ; You can adjust several parameters relating to the jitter buffer. ; The jitter buffer's function is to compensate for varying network delay. ; All the jitter buffer settings except dropcount are in milliseconds. ; The jitter buffer works for INCOMING audio - the outbound audio ; will be dejittered by the jitter buffer at the other end. ; jitterbuffer=no ; Whether you want the jitter buffer at all. ;dropcount=2; The jitter buffer is sized such that no more than dropcount ; frames would have been too late over the last 2 seconds. ; Set to a small number. 3 represents 1.5% of frames dropped ;maxjitterbuffer=500; A maximum size for the jitter buffer. Setting a reasonable maximum ; here will prevent the call delay from rising to silly values in ; extreme situations. ;maxexcessbuffer=80 ; If conditions improve after a period of high jitter, the jitter buffer ; can end up bigger than necessary. If it ends up more than ; maxexcessbuffer bigger than needed, Asterisk will start gradually ; decreasing the amount of jitter buffering. ;minexcessbuffer=80 ; Sets a desired mimimum amount of headroom in the jitter buffer. ; If Asterisk has less headroom than this, then it will start gradually ; increasing the amount of jitter buffering. ;jittershrinkrate=1 ; When the jitter buffer is being gradually shrunk (or enlarged), ; how many millisecs shall we take off per 20ms frame received? ; Use a small number, or you will be able to hear it changing. ; An example: if you set this to 2, then the jitter buffer size will ; change by 100 millisec per second. ;trunkfreq=20 ; How frequently to send trunk msgs (in ms) authdebug=no; You can disable authentication debugging to reduce ; the amount of debugging traffic. tos=lowdelay; You can set values for your TOS bits to help improve performance. ; Can be lowdelay, throughput, reliability, mincost or none. ;mailboxdetail=yes ; If set to yes, the user receives the actual ; new/old message counts, not just a yes/no as to ; whether they have messages. register = in-xxx##XxX#X:[EMAIL PROTECTED] ; ### PROVIDERS ### [voicepulse]; For inbound context=VPWS type=user host=gw5.voicepulse.com accountcode=1 [vpconnect-t01] ; For outbound type=peer secret=xXx##Xxx## host=gwiaxt01.voicepulse.com auth=md5 qualify=yes accountcode=1 [vpconnect-t02] ; Outbound backup type=peer secret=xXx##Xxx## host=gwiaxt02.voicepulse.com auth=md5 qualify=yes accountcode=1 -- extensions.conf -- [VPWS] ; All Inbound Voicepulse DID numbers go here ; From here it is distributed to the propper place ;; - Some Company - exten = 1235551212,1,Goto(company,1235551212,1) [company] ;
[Asterisk-Users] Harddisk noise on TE410P
Hi, I have this strange problem I need some help with.. It appears that I have harddisk noise captured by a Digium TE410P card (Same problem on 2 identical machines..) The machines are two Compaq Proliant DL320 G3's... Does anyone else have this problem.. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] which distro for asterisk?
We discussed this earlier and I believe the general consensus was that it's personal choice. I've personally used Asterisk on Redhat 9.0, Fedora Core 1 and Gentoo 2004.2 Each has required some minor securing and cleaning up, but Redhat/Fedora tended to need more babying as far as securing default configs and speeding up certain things. But after a little bit of time with it, I had it running efficient and Asterisk loved it. The main problem I found with Redhat/Fedora was the default kernel, the sources are all messed up and the zaptel/libpri drivers didn't compile quite right. I simply downloaded the latest kernel, compiled it (and set it to my specific hardware to conserve memory) and libpri/zaptel compiled fine. Personally I'm now using Gentoo. What I did with Gentoo was emerge asterisk-0.9.0 or whatever, and it handled all the dependencies for more. When I was done with that, I did a emerge cvs and once I had cvs I downloaded, compiled and upgraded to the latest libpri/zaptel/asterisk, just did a make upgrade. Gentoo tends to be a faster OS and more efficient with resources, it gives you bare minimum. Which sometimes is good, for security and such, but also a pain when it comes to standard interaction. ie by default they don't include ftp or telnet and traceroute just commands I'm used to having, nothing I can't emerge though. In the end, it's personal choice. For now I've gone with Gentoo. Asterisk really isn't that resource intensive, it seems to like memory a lot, but other than that I don't see it putting heavy loads on my systems, and the speed difference between two identical machines, one with with Fedora Core 1 and one with Gentoo, is almost imperceivable when it comes to working with Asterisk, (ie, the IVR and playback of messages, and interacting with the voicemail system, etc.). Tzafrir Cohen wrote: Hi I want to play a bit with Asterisk. I currentlly install a new system for that and I would like to get your recommendations regarding the linux distro to use there. This is NOT intended to become a general distro flame war. My favorite distro is and no argument that you flame will convince me here (probably because I've heard it before). However I would like to minimize the OS maintinance task. I really wouldn't like to start worrying about upgrading sshd due to some stupid secuirty hole, and to worry what will it break on my system. I expect my distro to do that for me. I'd also like to have solid astrisk packages that won't break unnecessarily when the sshd package is updated next time. Hopefully also some sort of integration of zaptel in the distro's kernel package. I saw numerous complaints about unofficial RPM packages of asterisk. Besides them, the following free distros include asterisk packages: 1. Debian: http://packages.debian.org/asterisk . 2. Gentoo: Current package seems to be version 0.9.0 from 10-May-2004 3. The DAG repository for RH/Fedora: http://dag.wieers.com/packages/asterisk/ I have some experince with Debian, Mandrake and RedHat/Fedora. I'm unfamiliar with Gentoo and I have no good/bad experince with DAG packages with respect to quality and stability. Any recommendations, relevant experince and other learned opinions? thx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Harddisk noise on TE410P
You can actually hear the hard drive noise when calling out or receiving a call? A clicking sound, or like an electrical noise? I doubt this is being done through the motherboard, how close is the card to the power supply and/or the power wires going into the hard drives? Are they less (or more) shielded than normal? Have you tried it in a non-Compaq proliant server? Claus Futtrup wrote: Hi, I have this strange problem I need some help with.. It appears that I have harddisk noise captured by a Digium TE410P card (Same problem on 2 identical machines..) The machines are two Compaq Proliant DL320 G3's... Does anyone else have this problem.. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] incomming call rejected using IAX2 with FWD
What does your host= line show in the iax.conf for fwd? I found that iax.conf hates it when you use host=x.x.x.x so instead I had to use host=dynamic and defaultip=x.x.x.x or something like that. It's very finicky. Storm D. J. Petersen wrote: Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect attempt from 65.39.205.121 Any ideas? Thanks, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie question about PBX Call Pickup
Hi, sorry for annoying question; i read http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup without understanding: 1. how to add an ext. to a pickup group (ie:. how to populate pickup group) 2. how 'Directed pickup' does work? You dial the pickup number and your extension, and the call will only transfer if it is your extension should i digit something like '*8, then dial my extension? i tried to dial my extension but i got a busy tone maurizio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: zaptel configuration
Asalamualaikum Atif, i saw your guy's ad in spider magazine. sounds cool... yeah, i got asterisk to work, i had to build zaptel before asterisk. just trying to transfer from one line to another now... thanks Imran Atif Rasheed wrote: well Imran, I am not a guru of Asterisk, but I think my suggestions might work, Hi, I've been trying to get my zaptel x100p cards working for the past week now. this is what I've done: first, this should have been done with Zaptel, not asterisk install Zaptel, like this: make clean make linux26 (if kernel version is 2.6 or above, else do 'make' only) make install installed asterisk: make clean make linux 26 (for fedora core 2) make install and, this should have been done for Asterisk i.e. install Asterisk, like: make clean make make install installed zaptel: make clean make make install then do 'modprobe's', 'ztcfg -v', and then do 'asterisk -vvc' then check for errors, if any did a modprobe zaptel, and wcfxo got this in /var/log/messages: PCI: found IRQ 11 for device :00:0f.0 wcfxo: daa mode is 'FCC' found a wildcard fxo: wildcard x101p ... in zaptel.conf: fxsks=1-2 in zapata.conf: signalling = fxs_ks channel = 1 channel = 2 yet when i run asterisk, the zap show channels command doesn't work. in a previous thread they mentioned this is because some chan_zap.so file isn't loaded because of the zaptel installation. I was told I had to REINSTALL asterisk after the zaptel stuff, which again didn't do anything. How can this be so hard to even get installed? Thanks, Imran hope, it will work this time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] which distro for asterisk?
I'm using it on Debian Stable (Woody), works great, using it with the backports.org 2.6.7-1-686-smp kernel, zaptel drivers compile fine with their headers. I think it's all a matter of personal preference. I prefer Debian, so I use it, use whatever you like best :) -Tim -Original Message- From: Deon Rodden [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] which distro for asterisk? We discussed this earlier and I believe the general consensus was that it's personal choice. I've personally used Asterisk on Redhat 9.0, Fedora Core 1 and Gentoo 2004.2 Each has required some minor securing and cleaning up, but Redhat/Fedora tended to need more babying as far as securing default configs and speeding up certain things. But after a little bit of time with it, I had it running efficient and Asterisk loved it. The main problem I found with Redhat/Fedora was the default kernel, the sources are all messed up and the zaptel/libpri drivers didn't compile quite right. I simply downloaded the latest kernel, compiled it (and set it to my specific hardware to conserve memory) and libpri/zaptel compiled fine. Personally I'm now using Gentoo. What I did with Gentoo was emerge asterisk-0.9.0 or whatever, and it handled all the dependencies for more. When I was done with that, I did a emerge cvs and once I had cvs I downloaded, compiled and upgraded to the latest libpri/zaptel/asterisk, just did a make upgrade. Gentoo tends to be a faster OS and more efficient with resources, it gives you bare minimum. Which sometimes is good, for security and such, but also a pain when it comes to standard interaction. ie by default they don't include ftp or telnet and traceroute just commands I'm used to having, nothing I can't emerge though. In the end, it's personal choice. For now I've gone with Gentoo. Asterisk really isn't that resource intensive, it seems to like memory a lot, but other than that I don't see it putting heavy loads on my systems, and the speed difference between two identical machines, one with with Fedora Core 1 and one with Gentoo, is almost imperceivable when it comes to working with Asterisk, (ie, the IVR and playback of messages, and interacting with the voicemail system, etc.). Tzafrir Cohen wrote: Hi I want to play a bit with Asterisk. I currentlly install a new system for that and I would like to get your recommendations regarding the linux distro to use there. This is NOT intended to become a general distro flame war. My favorite distro is and no argument that you flame will convince me here (probably because I've heard it before). However I would like to minimize the OS maintinance task. I really wouldn't like to start worrying about upgrading sshd due to some stupid secuirty hole, and to worry what will it break on my system. I expect my distro to do that for me. I'd also like to have solid astrisk packages that won't break unnecessarily when the sshd package is updated next time. Hopefully also some sort of integration of zaptel in the distro's kernel package. I saw numerous complaints about unofficial RPM packages of asterisk. Besides them, the following free distros include asterisk packages: 1. Debian: http://packages.debian.org/asterisk . 2. Gentoo: Current package seems to be version 0.9.0 from 10-May-2004 3. The DAG repository for RH/Fedora: http://dag.wieers.com/packages/asterisk/ I have some experince with Debian, Mandrake and RedHat/Fedora. I'm unfamiliar with Gentoo and I have no good/bad experince with DAG packages with respect to quality and stability. Any recommendations, relevant experince and other learned opinions? thx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse Connect DTMF with IAX2
I'm using RC2 and last weekend's changes from VoicePulse. Outbound calling and dtmf works fine. However, an inbound call to my DID cannot send dtmf digits to the IVR. Thoughts? Deon Rodden wrote: Here's my iax.conf and extensions.conf (I have not yet made the recent changes they just emailed about a day ago, this is twice in a two month period, jeesh) I have tested inbound and outbound dtmf. I use the g.711 codec and use inband. iax.conf -- [general] port=5036 bindaddr=0.0.0.0 context=incoming ;iaxcompat=yes ; Set iaxcompat to yes if you plan to use layered switches. ; It incurs a small performance hit to enable it. delayreject=yes ; For increased security against brute force password attacks. ; Enabling this will delay the sending of authentication ; reject for REGREQ or AUTHREP if there is a password. amaflags=documentation ; global default AMA flag for iaxtel calls. These flags ; are used in the generation of call detail records. ;accountcode=1 ; default account for Call Detail Records in addition ; to specifying on a per-user basis. language=en ; Global default language for users. ; If omitted, will fallback to english bandwidth=high ; Specify bandwidth of low, medium, or high to ; control which codecs are used in general. allow=all ; Which codecs to allow, same as bandwidth=high disallow=g723.1 ; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. ; You can adjust several parameters relating to the jitter buffer. ; The jitter buffer's function is to compensate for varying network delay. ; All the jitter buffer settings except dropcount are in milliseconds. ; The jitter buffer works for INCOMING audio - the outbound audio ; will be dejittered by the jitter buffer at the other end. ; jitterbuffer=no ; Whether you want the jitter buffer at all. ;dropcount=2; The jitter buffer is sized such that no more than dropcount ; frames would have been too late over the last 2 seconds. ; Set to a small number. 3 represents 1.5% of frames dropped ;maxjitterbuffer=500; A maximum size for the jitter buffer. Setting a reasonable maximum ; here will prevent the call delay from rising to silly values in ; extreme situations. ;maxexcessbuffer=80 ; If conditions improve after a period of high jitter, the jitter buffer ; can end up bigger than necessary. If it ends up more than ; maxexcessbuffer bigger than needed, Asterisk will start gradually ; decreasing the amount of jitter buffering. ;minexcessbuffer=80 ; Sets a desired mimimum amount of headroom in the jitter buffer. ; If Asterisk has less headroom than this, then it will start gradually ; increasing the amount of jitter buffering. ;jittershrinkrate=1 ; When the jitter buffer is being gradually shrunk (or enlarged), ; how many millisecs shall we take off per 20ms frame received? ; Use a small number, or you will be able to hear it changing. ; An example: if you set this to 2, then the jitter buffer size will ; change by 100 millisec per second. ;trunkfreq=20 ; How frequently to send trunk msgs (in ms) authdebug=no; You can disable authentication debugging to reduce ; the amount of debugging traffic. tos=lowdelay; You can set values for your TOS bits to help improve performance. ; Can be lowdelay, throughput, reliability, mincost or none. ;mailboxdetail=yes ; If set to yes, the user receives the actual ; new/old message counts, not just a yes/no as to ; whether they have messages. register = in-xxx##XxX#X:[EMAIL PROTECTED] ; ### PROVIDERS ### [voicepulse]; For inbound context=VPWS type=user host=gw5.voicepulse.com accountcode=1 [vpconnect-t01] ; For outbound type=peer secret=xXx##Xxx## host=gwiaxt01.voicepulse.com auth=md5 qualify=yes accountcode=1 [vpconnect-t02] ; Outbound backup type=peer secret=xXx##Xxx## host=gwiaxt02.voicepulse.com auth=md5 qualify=yes accountcode=1 -- extensions.conf
RE: [Asterisk-Users] which distro for asterisk?
I've been having troubles compiling in the openh323 on both redhat and debian... one of the biggest problems I had w/ Debian is it couldn't find alot of libraries like termcap etc... Has anyone else ran into these problems? -Original Message- From: Tim Jackson [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 10:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] which distro for asterisk? I'm using it on Debian Stable (Woody), works great, using it with the backports.org 2.6.7-1-686-smp kernel, zaptel drivers compile fine with their headers. I think it's all a matter of personal preference. I prefer Debian, so I use it, use whatever you like best :) -Tim -Original Message- From: Deon Rodden [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] which distro for asterisk? We discussed this earlier and I believe the general consensus was that it's personal choice. I've personally used Asterisk on Redhat 9.0, Fedora Core 1 and Gentoo 2004.2 Each has required some minor securing and cleaning up, but Redhat/Fedora tended to need more babying as far as securing default configs and speeding up certain things. But after a little bit of time with it, I had it running efficient and Asterisk loved it. The main problem I found with Redhat/Fedora was the default kernel, the sources are all messed up and the zaptel/libpri drivers didn't compile quite right. I simply downloaded the latest kernel, compiled it (and set it to my specific hardware to conserve memory) and libpri/zaptel compiled fine. Personally I'm now using Gentoo. What I did with Gentoo was emerge asterisk-0.9.0 or whatever, and it handled all the dependencies for more. When I was done with that, I did a emerge cvs and once I had cvs I downloaded, compiled and upgraded to the latest libpri/zaptel/asterisk, just did a make upgrade. Gentoo tends to be a faster OS and more efficient with resources, it gives you bare minimum. Which sometimes is good, for security and such, but also a pain when it comes to standard interaction. ie by default they don't include ftp or telnet and traceroute just commands I'm used to having, nothing I can't emerge though. In the end, it's personal choice. For now I've gone with Gentoo. Asterisk really isn't that resource intensive, it seems to like memory a lot, but other than that I don't see it putting heavy loads on my systems, and the speed difference between two identical machines, one with with Fedora Core 1 and one with Gentoo, is almost imperceivable when it comes to working with Asterisk, (ie, the IVR and playback of messages, and interacting with the voicemail system, etc.). Tzafrir Cohen wrote: Hi I want to play a bit with Asterisk. I currentlly install a new system for that and I would like to get your recommendations regarding the linux distro to use there. This is NOT intended to become a general distro flame war. My favorite distro is and no argument that you flame will convince me here (probably because I've heard it before). However I would like to minimize the OS maintinance task. I really wouldn't like to start worrying about upgrading sshd due to some stupid secuirty hole, and to worry what will it break on my system. I expect my distro to do that for me. I'd also like to have solid astrisk packages that won't break unnecessarily when the sshd package is updated next time. Hopefully also some sort of integration of zaptel in the distro's kernel package. I saw numerous complaints about unofficial RPM packages of asterisk. Besides them, the following free distros include asterisk packages: 1. Debian: http://packages.debian.org/asterisk . 2. Gentoo: Current package seems to be version 0.9.0 from 10-May-2004 3. The DAG repository for RH/Fedora: http://dag.wieers.com/packages/asterisk/ I have some experince with Debian, Mandrake and RedHat/Fedora. I'm unfamiliar with Gentoo and I have no good/bad experince with DAG packages with respect to quality and stability. Any recommendations, relevant experince and other learned opinions? thx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P Questions: Voicemail and Phone Port questions
Hello fellow * users, I've been experimenting like a madman lately with asterisk, and I just love it. Just reading this list and asking a few questions here and there has helped me out a great amount. Not to mention the excellent resource we call the Wiki. I have searched for answers to the two following questions, but couldn't find anything, it is entirely possible i was just searching for the wrong terms or something, if so please point me in the right direction. I recently purchased an X100P, and it's working great at home as a voicemail only box right now. Eventually I will be adding some FXS ports to support internal extensions. Now, my questions are this: 1. Currently, I have a default channel with the commands 'wait(12)' followed by 'answer' for the X100P in my extensions.conf. the problem with this is that if my sister or significant other picks up the phone during the call in process, asterisk does not stop it's 'wait' and then 'answer' of the line. Is there a way to make asterisk recognize that the line is not ringing anymore (thus, somebody must have picked up) and not go to the voicemail menu? (Without purchasing fxs ports) Is there something like 'AnswerOnlyIfRing' or something similar? 2. With only a X100P is there a way to 'disable' the second port on it? Currently I have my incoming PSTN line going into the X100P, then my internal phones are all 'daisychained' together coming out of the 'out' port on the X100P. What I'm looking for is a way to make it so that if somebody is leaving a message to us, and an inside line picks up, they don't hear the person leaving the message. All they hear is dead silence. Or music on hold or something. I realize for part two of this question i'm probably going to need at least one FXS port. but I wasn't sure if anyone out there had come up with some k-leet workaround :) Thanks for your time, Matt G ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] which distro for asterisk?
I had that problem, but apt-get install did the trick. On Tuesday 31 August 2004 02:53 pm, Huddleston, Robert wrote: I've been having troubles compiling in the openh323 on both redhat and debian... one of the biggest problems I had w/ Debian is it couldn't find alot of libraries like termcap etc... Has anyone else ran into these problems? -Original Message- From: Tim Jackson [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 10:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] which distro for asterisk? I'm using it on Debian Stable (Woody), works great, using it with the backports.org 2.6.7-1-686-smp kernel, zaptel drivers compile fine with their headers. I think it's all a matter of personal preference. I prefer Debian, so I use it, use whatever you like best :) -Tim -Original Message- From: Deon Rodden [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] which distro for asterisk? We discussed this earlier and I believe the general consensus was that it's personal choice. I've personally used Asterisk on Redhat 9.0, Fedora Core 1 and Gentoo 2004.2 Each has required some minor securing and cleaning up, but Redhat/Fedora tended to need more babying as far as securing default configs and speeding up certain things. But after a little bit of time with it, I had it running efficient and Asterisk loved it. The main problem I found with Redhat/Fedora was the default kernel, the sources are all messed up and the zaptel/libpri drivers didn't compile quite right. I simply downloaded the latest kernel, compiled it (and set it to my specific hardware to conserve memory) and libpri/zaptel compiled fine. Personally I'm now using Gentoo. What I did with Gentoo was emerge asterisk-0.9.0 or whatever, and it handled all the dependencies for more. When I was done with that, I did a emerge cvs and once I had cvs I downloaded, compiled and upgraded to the latest libpri/zaptel/asterisk, just did a make upgrade. Gentoo tends to be a faster OS and more efficient with resources, it gives you bare minimum. Which sometimes is good, for security and such, but also a pain when it comes to standard interaction. ie by default they don't include ftp or telnet and traceroute just commands I'm used to having, nothing I can't emerge though. In the end, it's personal choice. For now I've gone with Gentoo. Asterisk really isn't that resource intensive, it seems to like memory a lot, but other than that I don't see it putting heavy loads on my systems, and the speed difference between two identical machines, one with with Fedora Core 1 and one with Gentoo, is almost imperceivable when it comes to working with Asterisk, (ie, the IVR and playback of messages, and interacting with the voicemail system, etc.). Tzafrir Cohen wrote: Hi I want to play a bit with Asterisk. I currentlly install a new system for that and I would like to get your recommendations regarding the linux distro to use there. This is NOT intended to become a general distro flame war. My favorite distro is and no argument that you flame will convince me here (probably because I've heard it before). However I would like to minimize the OS maintinance task. I really wouldn't like to start worrying about upgrading sshd due to some stupid secuirty hole, and to worry what will it break on my system. I expect my distro to do that for me. I'd also like to have solid astrisk packages that won't break unnecessarily when the sshd package is updated next time. Hopefully also some sort of integration of zaptel in the distro's kernel package. I saw numerous complaints about unofficial RPM packages of asterisk. Besides them, the following free distros include asterisk packages: 1. Debian: http://packages.debian.org/asterisk . 2. Gentoo: Current package seems to be version 0.9.0 from 10-May-2004 3. The DAG repository for RH/Fedora: http://dag.wieers.com/packages/asterisk/ I have some experince with Debian, Mandrake and RedHat/Fedora. I'm unfamiliar with Gentoo and I have no good/bad experince with DAG packages with respect to quality and stability. Any recommendations, relevant experince and other learned opinions? thx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list
[Asterisk-Users] Re: Newbie - Voicemail Password Help
- Original Message -
Hello All.
I'm just beginning with Asterisk and I have it all working now. I'm using
Asterisk 1.0 RC1.
My only question is this; when I check my voice mail the PBX simply says
password. I wanted to make it say please enter your voice mail
password so
I am using Background(pls-enter-vm-password).
However now I hear Please enter your voice mail password password when I
check my messages.
That's not a type-o. It says password twice.
Here is my extensions.conf file.
[macro-vmanswer]
exten = s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
exten = s,2,Background(pls-enter-vm-password)
exten = s,3,VoicemailMain(${ARG1})
exten = s,4,Hangup
exten = s,5,Voicemail(u${ARG1})
exten = s,6,Hangup
try
exten = xxx,1,VoicemailMain(${CALLERIDNUM)
exten = xxx,2,Hangup
for your voicemailmain extension. this will recognize that your callerid if
you have a mailbox on the system. note that your mailbox and your caller id
must match
alternatively, you could go into the /var/lib/asterisk/sounds directory and
rename the file vm-password to old.vm-password then rename your file
pls-enter-vmpassword to vm-password
that way you would not have to alter the code at all.
[default]
exten = 1002,1,Macro(vmanswer,1002)
The whole point of the vmanswer macro is to go to the voice mail main menu
automatically when calling from your own phone, otherwise it sends callers
to
the voice mail system to leave a message. Perhaps there's a better way to
do
this as well. If so, please let me know.
Regards,
Paul
Good Luck
Jason Kawakami
www.optellabs.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SoundPoint... Gains - Which is for speakerphone
On Mon, 30 Aug 2004 22:09:26 -0500, John Baker wrote: Hmmm... Hands Free might be: voice.gain.rx.digital.chassis=15 (15 is my setting) Call waiting? You can turn it off in sip.cfg - do not disturb settings I think. Don't know about gain for call waiting. You might try playing with some of the variables in ipmid.cfg under ringType John Is there some way to get the phones current settings extracted to a sample xml file? The sample files that come with the software distributions are just empty frameworks with no settings. In my case I have some IP600s. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 It's a funny thing about life; if you refuse to accept anything but the best, you quite often get it. W. Somerset Maugham ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] which distro for asterisk?
On Tue, Aug 31, 2004 at 11:02:30AM +, Brian Wilkins wrote: I had that problem, but apt-get install did the trick. Not to mention apt-get source and apt-get build-dep if you need to patch existing packages -- Tzafrir Cohen +---+ http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend| mailto:[EMAIL PROTECTED] +---+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SoundPoint... Gains - Which is for speakerphone
1) The samples are empty? No, they have variables with settings. Maybe I'm not understanding you. 2) I don't know how to dump the current settings to an xml file. You might try increasing the log level, but I doubt you're going to get a pretty looking xml file written to the log files. You're better off messing with the included config files. John Michael Graves wrote: On Mon, 30 Aug 2004 22:09:26 -0500, John Baker wrote: Hmmm... Hands Free might be: voice.gain.rx.digital.chassis=15 (15 is my setting) Call waiting? You can turn it off in sip.cfg - do not disturb settings I think. Don't know about gain for call waiting. You might try playing with some of the variables in ipmid.cfg under ringType John Is there some way to get the phones current settings extracted to a sample xml file? The sample files that come with the software distributions are just empty frameworks with no settings. In my case I have some IP600s. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 It's a funny thing about life; if you refuse to accept anything but the best, you quite often get it. ? W. Somerset Maugham ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 300 - Displaying Only Caller NAME... What about NUMBER?
I got most of the features of my phone working. Polycom TEch support refuses to help or even talk to me. So I'll have to ask here again. On incoming calls, only the NAME is displayed. I am trying to figure out how to get the NAME NUMBER displayed. If anyone can help me do this it would be GREATLY appreciated. Thank you in advance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint... Gains - Whichis for speakerphone
If that was possible, that would make my life easier as well :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Tuesday, August 31, 2004 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom SoundPoint... Gains - Whichis for speakerphone On Mon, 30 Aug 2004 22:09:26 -0500, John Baker wrote: Hmmm... Hands Free might be: voice.gain.rx.digital.chassis=15 (15 is my setting) Call waiting? You can turn it off in sip.cfg - do not disturb settings I think. Don't know about gain for call waiting. You might try playing with some of the variables in ipmid.cfg under ringType John Is there some way to get the phones current settings extracted to a sample xml file? The sample files that come with the software distributions are just empty frameworks with no settings. In my case I have some IP600s. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 It's a funny thing about life; if you refuse to accept anything but the best, you quite often get it. - W. Somerset Maugham ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint... Gains -Which is for speakerphone
John, By chance do you know how to set a default ringer? What I have done is the following: DEFAULT se.rt.1.name=Default se.rt.1.type=ring se.rt.1.ringer=7 se.rt.1.callWait=6 se.rt.1.mod=1/ As you can see, I want 7 to be the default ringer for line 1... For some reason, it doesn't take these changes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Tuesday, August 31, 2004 11:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom SoundPoint... Gains -Which is for speakerphone 1) The samples are empty? No, they have variables with settings. Maybe I'm not understanding you. 2) I don't know how to dump the current settings to an xml file. You might try increasing the log level, but I doubt you're going to get a pretty looking xml file written to the log files. You're better off messing with the included config files. John Michael Graves wrote: On Mon, 30 Aug 2004 22:09:26 -0500, John Baker wrote: Hmmm... Hands Free might be: voice.gain.rx.digital.chassis=15 (15 is my setting) Call waiting? You can turn it off in sip.cfg - do not disturb settings I think. Don't know about gain for call waiting. You might try playing with some of the variables in ipmid.cfg under ringType John Is there some way to get the phones current settings extracted to a sample xml file? The sample files that come with the software distributions are just empty frameworks with no settings. In my case I have some IP600s. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 It's a funny thing about life; if you refuse to accept anything but the best, you quite often get it. ? W. Somerset Maugham ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 300 - Displaying Only Caller NAME... What about NUMBER?
Put a NoOp(Caller*ID is ${CALLERID}) in your dialplan JUST before the
Dial to the Polycom. See if the correct name and number shows up on the
console when the NoOp runs. If it does, there's a problem in the
Polycom, if there is no NAME then you have a problem with your Asterisk
config.
On Tue, 2004-08-31 at 10:08, Matthew Marlowe wrote:
I got most of the features of my phone working. Polycom TEch support
refuses to help or even talk to me. So I'll have to ask here again.
On incoming calls, only the NAME is displayed. I am trying to figure
out how to get the NAME NUMBER displayed.
If anyone can help me do this it would be GREATLY appreciated.
Thank you in advance
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can i send calling costs to a SIP IP phone display
Is there a solution for asterisk to send the calling costs to a display of a grandstream Bt101 phone. Does anybody know if there is a solution for this? Greetings Han ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] answer from wrong port
Hi everyone, I'm having a little problem and was wondering whether anyone would have any ideas or pointers for me. I've been working on load-balancing asterisk and have had a pretty successful setup using LVS and IP tunneling (plus a bit of iptables nating). I am only load balancing the SIP registration while the RTP between the SIP phone and the asterisk server and between the asterisk server and the CISCO AS5300 is being done directly with the real IP. Now this setup worked wonderfully, and I had tested with SIP phones behind different routers to see if Natting wasn't causing a problem and everything worked fine. But one of my locations recently changed routers (Linksys WRT54G) and the SIP phone no longer registers with the asterisk servers. After a bit of sniffing adn testing, here's what I came up with. If the phone connects directly to the asterisk server without load-balancing, it works fine. If the phone connects to the asterisk server through the load-balancing, the REGISTER packet comes into the asterisk server, but the reply instead of being sent-out from source-port 5060, it's sent out from source-port 1343 (or other lowest free port (1024,1026) and is blocked at the linksys gateway. Any ideas why asterisk doesn't use the 5060 source port in the reply? I'm unfortunately using version 0.9.0 of asterisk (my boss doesn't want to go with CVS). P.S. The iptables part of the load-balancing NATs the source IP of the reply packets as being from the virtual IP because asterisk sets it as from the real IP. The rest is normal lvs Thanks for any help Benjamin -- \\\|/// \\ - - // ( @ @ ) ---oOOo-(_)-oOOo--- There are times when truth is stranger than fiction and lunch time is one of them. --Oooo- oooO( ) Benjamin Benthos Lawetz ( ) ) /mailto:[EMAIL PROTECTED] \ ((_/ ICQ# 4269530 \_) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 300 - Displaying Only CallerNAME... What about NUMBER?
I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up
fine on my 7960... W/ the name on top and the number below that.
-- Executing NoOp(SIP/614-3ede, Caller*ID is Matthew Marlowe
6092521155) in new stack
When the phone rings, only 'Matthew Marlowe' would display. When I
answer, both the Name Number will show. It's simple while the phone
is ringing that it doesn't display.
I mean I doubt the polycom is malfunctioning, that's why I think there
might be some configuration to change... But what, I have no idea.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Tuesday, August 31, 2004 11:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP 300 - Displaying Only
CallerNAME... What about NUMBER?
Put a NoOp(Caller*ID is ${CALLERID}) in your dialplan JUST before the
Dial to the Polycom. See if the correct name and number shows up on the
console when the NoOp runs. If it does, there's a problem in the
Polycom, if there is no NAME then you have a problem with your Asterisk
config.
On Tue, 2004-08-31 at 10:08, Matthew Marlowe wrote:
I got most of the features of my phone working. Polycom TEch support
refuses to help or even talk to me. So I'll have to ask here again.
On incoming calls, only the NAME is displayed. I am trying to figure
out how to get the NAME NUMBER displayed.
If anyone can help me do this it would be GREATLY appreciated.
Thank you in advance
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a
related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 300 - Displaying Only CallerNAME... What about NUMBER?
On Tue, 2004-08-31 at 10:37, Matthew Marlowe wrote: I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up fine on my 7960... W/ the name on top and the number below that. -- Executing NoOp(SIP/614-3ede, Caller*ID is Matthew Marlowe 6092521155) in new stack When the phone rings, only 'Matthew Marlowe' would display. When I answer, both the Name Number will show. It's simple while the phone is ringing that it doesn't display. Is Matthew Marlowe in the Polycom directory application? Is so that might be the reason it's not working as expected. I seem to recall reading about it somwewhere in the Admin guide in the section about the on phone directory/speed dial list -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Harddisk noise on TE410P
Claus- This is a problem that interests me, as I'm about to deploy TEN of these at a customer site, all with TE410P's. I'm currently load testing one Proliant box (3GHz P4 processor) looping 59 calls out to 59 calls in (leaving one channel open) - ie: lots of load. While I'm doing this, I call in from another asterisk box over IAX, route this call out over a TE410 channel and back in, and listen to a prompt. I don't hear any unusual noise, and the box is performing well otherwise. Please supply more detail: What kind of disk, which Linux distro - and, what is the noise you're hearing? Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup Sent: Tuesday, August 31, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Harddisk noise on TE410P Hi, I have this strange problem I need some help with.. It appears that I have harddisk noise captured by a Digium TE410P card (Same problem on 2 identical machines..) The machines are two Compaq Proliant DL320 G3's... Does anyone else have this problem.. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
This is nothing to do with SIP. It is an RTP issue, common to everything which uses RTP - SIP and H.323 included. I have been reading the RFCs and I'm a bit more familiar with how it works now although the algorithms are a bit over my head. I am somewhat new to RTP/VoIP, but I have a strong telecom/networking background so it makes things a bit easier to understand since they share a lot of common features.. I just thought from the post mentioning only IAX2 and some of the other codecs that SIP et. al. would be ignored... Sending no packets is perfectly valid, and normal, in RTP. If the receiving end takes no packets (other than, perhaps, an extremely long silence) as a disconnect it does not comply with the RTP spec. DTX is much despised, and CNG only slightly better. They just sound good (pun intende) on paper. While I realize that hanging up on silence is not a desired behavior, unfortunately lots of things are out of spec... Look at Cisco's POE implementation for example, it's completely reversed from 802.3af specs... If * had at least some kind of continuous CNG capability it would help in these situations... Silence should be acceptable and even desired because it saves bandwidth, but apparently some people (and switches) find it uncomfortable... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoicePulse Connect DTMF with IAX2
I'm using RC2 and last weekend's changes from VoicePulse. Outbound calling and dtmf works fine. However, an inbound call to my DID cannot send dtmf digits to the IVR. Thoughts? I have the same problem...my iax.conf is set up exactly as recommended per the recent Voicepulse changes and the configs they sent - my CVS is 7/14/04. Both inbound and outbound calling work, but no DTMF received on inbound calls. I found a post on the broadband reports forums regarding this issue, there where a few people who thought that it may affect VP customers who signed up for a DID in a new VP rate center/exchange...for example I've been waiting for VP to offer Colorado DID's (303 or 720) for quite awhile...so when I saw that they were available recently, I jumped on it and ordered one...so this a fairly new area code for them and I have the DTMF problem. I read other people that signed up for a fairly new area code having the same problem and emailing VP support to get it straightened out... I myself have sent them an email which they say they are checking into...I will be sure to let people know what my findings are. Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-users] PLC (Packet loss cancel) questions
I have been reading the RFCs and I'm a bit more familiar with how it works now although the algorithms are a bit over my head. I am somewhat new to RTP/VoIP, but I have a strong telecom/networking background so it makes things a bit easier to understand since they share a lot of common features.. I just thought from the post mentioning only IAX2 and some of the other codecs that SIP et. al. would be ignored... OOPS I meant... * protocols that SIP et. al. would be ignored... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse Connect DTMF with IAX2
On Tue, Aug 31, 2004 at 10:15:02AM -0400, Deon Rodden wrote:
exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,3,Congestion
This would dial the number twice..?
My config is
exten = _9.,1,Dial,IAX2/voicepulse/011${EXTEN:1}
exten = _9.,2,GotoIf($[ ${DIALSTATUS} != CONGESTION ${DIALSTATUS} != CHANUNAVAIL
]?6)
exten = _9.,3,Dial,IAX2/voicepulse2/011${EXTEN:1}
exten = _9.,4,GotoIf($[ ${DIALSTATUS} != CONGESTION ${DIALSTATUS} != CHANUNAVAIL
]?6)
exten = _9.,5,Congestion
exten = _9.,6,Hangup
arkadi.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limit the length of extensions
You limit them by context. You put your outbound dialing patterns in a
context that inbound callers cann't access.
Lyle
- Original Message -
From: Deon Rodden [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 9:05 AM
Subject: [Asterisk-Users] limit the length of extensions
How do I limit the length of an extension? In my test IVR/Automated
Attendant (whatever it's called), at the beginning it plays if you know
your parties 3 digit extension, you may enter it now) and then it gives
a list of options. If the caller puts the 3 digit extension, it goes
through fine, if they press 1, or 2 it goes to the selected menu option,
but if they dial 91235551212 it dials that phone number. Which of
course, is a big security risk.
Is there a way to limit the length of an extension for an incoming call?
My only solution right now is to duplicate ever single extension (about
50 of them) in a seperate context, one that does not have the _9.
extension in it, and then make the call in menu have access to that
context. However, if I put a limit in the entire context of 3 digits,
then my coworkers who's phones are in that context can only dial each
other, not 9 and an outside number. So it has to be an incoming limit or
something.
Another possibly creative solution would be to SetGroup(outsidecaller)
on the incoming line and then just before my outbound extension put
SetGroup(outsidecaller) and then a CheckGroup(2) or something like
that. I'd have to put another SetGroup in the outbound extension
because there's no way to specify the group with the checkgroup command,
it gets it from the setgroup statement.
Any help would be appreciated.
Thanks,
Deon
[incoming]
exten = 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
exten = 9543340726,2,setcidname(Blocked)
exten = 9543340726,3,setcidnum(00)
exten = 9543340726,4,Goto(companyname,beginmenu,1)
[companyname] ; All the phones, including outbound extensions are in
this context
exten = beginmenu,1,SetVar(CALLEDNAME=CompanyName)
exten = beginmenu,2,Wait,1
exten = beginmenu,3,Background(company-main)
exten = beginmenu,4,Background(ifyouknow)
exten = beginmenu,5,Goto(company_mainmenu,s,1)
exten =
_9NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
exten = 502,1,Dial(SIP/whatever1SIP/whatever2|30|m)
...
[company_mainmenu]
exten = s,1,Background(company-nav1)
exten = 1,1,Goto(company_sales,s,1) ; Sales
exten = 2,1,Goto(companyname,502,1) ; Accounting
exten = 3,1,Goto(companyname,508,1) ; Customer Care
exten = 4,1,Goto(companyname,507,1) ; Technical Support
exten = 5,1,Goto(companyname,202,1) ; Human Resources
exten = 6,1,Goto(companyname,202,1) ; Provisioning
exten = 7,1,Goto(companyname,214,1) ; Marketing
exten = 0,1,Goto(companyname,210,1) ; Operator
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN noob question
After reading a retarded amount of docs I'm still unable to figure out how to get Asterisk to monitor my phone line and pick it up when the phone rings...Im using a voice/fax/data modem on ttyS2. Any tips/pointers to another stack of docs? Is this even doable without special hardware? TIA, Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse Connect DTMF with IAX2
My DID is 303 as well. Marty Mastera wrote: I'm using RC2 and last weekend's changes from VoicePulse. Outbound calling and dtmf works fine. However, an inbound call to my DID cannot send dtmf digits to the IVR. Thoughts? I have the same problem...my iax.conf is set up exactly as recommended per the recent Voicepulse changes and the configs they sent - my CVS is 7/14/04. Both inbound and outbound calling work, but no DTMF received on inbound calls. I found a post on the broadband reports forums regarding this issue, there where a few people who thought that it may affect VP customers who signed up for a DID in a new VP rate center/exchange...for example I've been waiting for VP to offer Colorado DID's (303 or 720) for quite awhile...so when I saw that they were available recently, I jumped on it and ordered one...so this a fairly new area code for them and I have the DTMF problem. I read other people that signed up for a fairly new area code having the same problem and emailing VP support to get it straightened out... I myself have sent them an email which they say they are checking into...I will be sure to let people know what my findings are. Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limit the length of extensions
If I put my outbound rules in a different context, and then include
them in my main context, callers who call in will be able to access the
extensions in the main context, but not the included (ie the outbound
extensions) extensions called from the outbound context?
Lyle Giese wrote:
You limit them by context. You put your outbound dialing patterns in a
context that inbound callers cann't access.
Lyle
- Original Message -
From: Deon Rodden [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 9:05 AM
Subject: [Asterisk-Users] limit the length of extensions
How do I limit the length of an extension? In my test IVR/Automated
Attendant (whatever it's called), at the beginning it plays if you know
your parties 3 digit extension, you may enter it now) and then it gives
a list of options. If the caller puts the 3 digit extension, it goes
through fine, if they press 1, or 2 it goes to the selected menu option,
but if they dial 91235551212 it dials that phone number. Which of
course, is a big security risk.
Is there a way to limit the length of an extension for an incoming call?
My only solution right now is to duplicate ever single extension (about
50 of them) in a seperate context, one that does not have the _9.
extension in it, and then make the call in menu have access to that
context. However, if I put a limit in the entire context of 3 digits,
then my coworkers who's phones are in that context can only dial each
other, not 9 and an outside number. So it has to be an incoming limit or
something.
Another possibly creative solution would be to SetGroup(outsidecaller)
on the incoming line and then just before my outbound extension put
SetGroup(outsidecaller) and then a CheckGroup(2) or something like
that. I'd have to put another SetGroup in the outbound extension
because there's no way to specify the group with the checkgroup command,
it gets it from the setgroup statement.
Any help would be appreciated.
Thanks,
Deon
[incoming]
exten = 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
exten = 9543340726,2,setcidname(Blocked)
exten = 9543340726,3,setcidnum(00)
exten = 9543340726,4,Goto(companyname,beginmenu,1)
[companyname] ; All the phones, including outbound extensions are in
this context
exten = beginmenu,1,SetVar(CALLEDNAME=CompanyName)
exten = beginmenu,2,Wait,1
exten = beginmenu,3,Background(company-main)
exten = beginmenu,4,Background(ifyouknow)
exten = beginmenu,5,Goto(company_mainmenu,s,1)
exten =
_9NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
exten = 502,1,Dial(SIP/whatever1SIP/whatever2|30|m)
...
[company_mainmenu]
exten = s,1,Background(company-nav1)
exten = 1,1,Goto(company_sales,s,1) ; Sales
exten = 2,1,Goto(companyname,502,1) ; Accounting
exten = 3,1,Goto(companyname,508,1) ; Customer Care
exten = 4,1,Goto(companyname,507,1) ; Technical Support
exten = 5,1,Goto(companyname,202,1) ; Human Resources
exten = 6,1,Goto(companyname,202,1) ; Provisioning
exten = 7,1,Goto(companyname,214,1) ; Marketing
exten = 0,1,Goto(companyname,210,1) ; Operator
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Harddisk noise on TE410P
Hi there, The disks are SCSI Raid hotswap disks 1 RPM, P4 2.8 gig CPU, 1 Gig. of ram., and the server is running Red Hat 9.0. The sound is just like hearing a disk just muffled (sounds like strange static).. If you have a number I can call you at then you can hear it yourself. Kind Regards Claus Futtrup This message is for the designated recipient only and may contain privileged or confidential information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited. - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: 'Claus Futtrup' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 5:50 PM Subject: RE: [Asterisk-Users] Harddisk noise on TE410P Claus- This is a problem that interests me, as I'm about to deploy TEN of these at a customer site, all with TE410P's. I'm currently load testing one Proliant box (3GHz P4 processor) looping 59 calls out to 59 calls in (leaving one channel open) - ie: lots of load. While I'm doing this, I call in from another asterisk box over IAX, route this call out over a TE410 channel and back in, and listen to a prompt. I don't hear any unusual noise, and the box is performing well otherwise. Please supply more detail: What kind of disk, which Linux distro - and, what is the noise you're hearing? Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup Sent: Tuesday, August 31, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Harddisk noise on TE410P Hi, I have this strange problem I need some help with.. It appears that I have harddisk noise captured by a Digium TE410P card (Same problem on 2 identical machines..) The machines are two Compaq Proliant DL320 G3's... Does anyone else have this problem.. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN noob question
After reading a retarded amount of docs I'm still unable to figure out how to get Asterisk to monitor my phone line and pick it up when the phone rings...Im using a voice/fax/data modem on ttyS2. Any tips/pointers to another stack of docs? Is this even doable without special hardware? No, its not possible with the modem you're talking about. Interfaces to a telephone line require an FXO interface, which can take the form of digium's hardware products (www.digium.com), external (ethernet attached) modules (1204 from www.mediatrix.com), ISDN adapters, etc. If you have a broadband internet connection, you can also sign up with several different providers that provide telephone numbers in many cities, extending those numbers to your asterisk box across your broadband internet service. Probably the least expensive what to play with asterisk is to purchase the x100p card from digium (supports one telephone line). You might dig around the http://www.voip-info.org/tiki-index.php (wiki) as there is a substantial amount of information on that site. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Harddisk noise on TE410P
Claus: One difference is that I'm using the slower ATA disk, not the SCSI. Is the noise rhythmic (periodic) or constant? If periodic, what is the time between noise bursts? Do you hear the noise throughout a call, or just occasionally? Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup Sent: Tuesday, August 31, 2004 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Harddisk noise on TE410P Hi there, The disks are SCSI Raid hotswap disks 1 RPM, P4 2.8 gig CPU, 1 Gig. of ram., and the server is running Red Hat 9.0. The sound is just like hearing a disk just muffled (sounds like strange static).. If you have a number I can call you at then you can hear it yourself. Kind Regards Claus Futtrup This message is for the designated recipient only and may contain privileged or confidential information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited. - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: 'Claus Futtrup' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 5:50 PM Subject: RE: [Asterisk-Users] Harddisk noise on TE410P Claus- This is a problem that interests me, as I'm about to deploy TEN of these at a customer site, all with TE410P's. I'm currently load testing one Proliant box (3GHz P4 processor) looping 59 calls out to 59 calls in (leaving one channel open) - ie: lots of load. While I'm doing this, I call in from another asterisk box over IAX, route this call out over a TE410 channel and back in, and listen to a prompt. I don't hear any unusual noise, and the box is performing well otherwise. Please supply more detail: What kind of disk, which Linux distro - and, what is the noise you're hearing? Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup Sent: Tuesday, August 31, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Harddisk noise on TE410P Hi, I have this strange problem I need some help with.. It appears that I have harddisk noise captured by a Digium TE410P card (Same problem on 2 identical machines..) The machines are two Compaq Proliant DL320 G3's... Does anyone else have this problem.. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Newbie - Voicemail Password Help
Thank you!
I took your advise and replaced the original vm-password.gsm file. Worked like
a charm.
Thanks again,
Paul
--- Jason Kawakami [EMAIL PROTECTED] wrote:
- Original Message -
Hello All.
I'm just beginning with Asterisk and I have it all working now. I'm using
Asterisk 1.0 RC1.
My only question is this; when I check my voice mail the PBX simply says
password. I wanted to make it say please enter your voice mail
password so
I am using Background(pls-enter-vm-password).
However now I hear Please enter your voice mail password password when I
check my messages.
That's not a type-o. It says password twice.
Here is my extensions.conf file.
[macro-vmanswer]
exten = s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
exten = s,2,Background(pls-enter-vm-password)
exten = s,3,VoicemailMain(${ARG1})
exten = s,4,Hangup
exten = s,5,Voicemail(u${ARG1})
exten = s,6,Hangup
try
exten = xxx,1,VoicemailMain(${CALLERIDNUM)
exten = xxx,2,Hangup
for your voicemailmain extension. this will recognize that your callerid if
you have a mailbox on the system. note that your mailbox and your caller id
must match
alternatively, you could go into the /var/lib/asterisk/sounds directory and
rename the file vm-password to old.vm-password then rename your file
pls-enter-vmpassword to vm-password
that way you would not have to alter the code at all.
[default]
exten = 1002,1,Macro(vmanswer,1002)
The whole point of the vmanswer macro is to go to the voice mail main menu
automatically when calling from your own phone, otherwise it sends callers
to
the voice mail system to leave a message. Perhaps there's a better way to
do
this as well. If so, please let me know.
Regards,
Paul
Good Luck
Jason Kawakami
www.optellabs.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Do you Yahoo!?
Win 1 of 4,000 free domain names from Yahoo! Enter now.
http://promotions.yahoo.com/goldrush
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limit the length of extensions
You should be able to do that, but of course always test, test, test to make
sure.
Lyle
- Original Message -
From: Deon Rodden [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 11:24 AM
Subject: Re: [Asterisk-Users] limit the length of extensions
If I put my outbound rules in a different context, and then include
them in my main context, callers who call in will be able to access the
extensions in the main context, but not the included (ie the outbound
extensions) extensions called from the outbound context?
Lyle Giese wrote:
You limit them by context. You put your outbound dialing patterns in a
context that inbound callers cann't access.
Lyle
- Original Message -
From: Deon Rodden [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 9:05 AM
Subject: [Asterisk-Users] limit the length of extensions
How do I limit the length of an extension? In my test IVR/Automated
Attendant (whatever it's called), at the beginning it plays if you know
your parties 3 digit extension, you may enter it now) and then it gives
a list of options. If the caller puts the 3 digit extension, it goes
through fine, if they press 1, or 2 it goes to the selected menu option,
but if they dial 91235551212 it dials that phone number. Which of
course, is a big security risk.
Is there a way to limit the length of an extension for an incoming call?
My only solution right now is to duplicate ever single extension (about
50 of them) in a seperate context, one that does not have the _9.
extension in it, and then make the call in menu have access to that
context. However, if I put a limit in the entire context of 3 digits,
then my coworkers who's phones are in that context can only dial each
other, not 9 and an outside number. So it has to be an incoming limit or
something.
Another possibly creative solution would be to SetGroup(outsidecaller)
on the incoming line and then just before my outbound extension put
SetGroup(outsidecaller) and then a CheckGroup(2) or something like
that. I'd have to put another SetGroup in the outbound extension
because there's no way to specify the group with the checkgroup command,
it gets it from the setgroup statement.
Any help would be appreciated.
Thanks,
Deon
[incoming]
exten = 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
exten = 9543340726,2,setcidname(Blocked)
exten = 9543340726,3,setcidnum(00)
exten = 9543340726,4,Goto(companyname,beginmenu,1)
[companyname] ; All the phones, including outbound extensions are in
this context
exten = beginmenu,1,SetVar(CALLEDNAME=CompanyName)
exten = beginmenu,2,Wait,1
exten = beginmenu,3,Background(company-main)
exten = beginmenu,4,Background(ifyouknow)
exten = beginmenu,5,Goto(company_mainmenu,s,1)
exten =
_9NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
exten = 502,1,Dial(SIP/whatever1SIP/whatever2|30|m)
...
[company_mainmenu]
exten = s,1,Background(company-nav1)
exten = 1,1,Goto(company_sales,s,1) ; Sales
exten = 2,1,Goto(companyname,502,1) ; Accounting
exten = 3,1,Goto(companyname,508,1) ; Customer Care
exten = 4,1,Goto(companyname,507,1) ; Technical Support
exten = 5,1,Goto(companyname,202,1) ; Human Resources
exten = 6,1,Goto(companyname,202,1) ; Provisioning
exten = 7,1,Goto(companyname,214,1) ; Marketing
exten = 0,1,Goto(companyname,210,1) ; Operator
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limit the length of extensions
Why are you including your outbound context into your incoming context in
the first place? That doesn't make any sense?
I'm guessing that because you're using a number in your exten = you're
using an IP channel like SIP or H323? Is this correct? If you're using a
T1/PRI or POTS lines you need to use 's'.
Using your example, your dialplan should look something like this...
[incoming]
exten = 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
exten = 9543340726,2,setcidname(Blocked)
exten = 9543340726,3,setcidnum(00)
exten = 9543340726,4,Goto(companyname,beginmenu,1)
[companyname]
exten = beginmenu,1,SetVar(CALLEDNAME=CompanyName)
exten = beginmenu,2,Wait,1
exten = beginmenu,3,Answer() ; Answer the channel!
exten = beginmenu,4,Background(company-main)
exten = beginmenu,5,Background(ifyouknow)
exten = beginmenu,6,Goto(company_mainmenu,s,1)
exten = 502,1,Dial(SIP/whoever1SIP/whoever2sip/whoever3,30,m)
exten = 507,1,Dial(SIP/daveSIP/jimSIP/lisa,30,m)
...
[company_mainmenu]
exten = s,1,Background(company-nav1)
exten = 1,1,Goto(company_sales,s,1) ; Sales
exten = 2,1,Goto(companyname,502,1) ; Accounting
exten = 3,1,Goto(companyname,508,1) ; Customer Care
exten = 4,1,Goto(companyname,507,1) ; Technical Support
exten = 5,1,Goto(companyname,202,1) ; Human Resources
exten = 6,1,Goto(companyname,202,1) ; Provisioning
exten = 7,1,Goto(companyname,214,1) ; Marketing
exten = 0,1,Goto(companyname,210,1) ; Operator
...
Instead of jumping back and forth like this, I'd use macros to try and
condense the dialplan a bit...
I can help you more with this if you'd like...
Then for people inside the company there's this...
[outbound-local]
exten = _9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,t) ; for
7-digit dialing
exten = _91800NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _91888NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _91877NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _91866NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
[outbound-ld]
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
[outbound-international]
exten = _9011.,Dial(SIP${EXTEN:[EMAIL PROTECTED],60,T)
[office]
include = outbound-local
include = outbound-ld
include = outbound-international
exten = _[1-5]XX,1,Dial(SIP/${EXTEN},25,tT) ; This is assuming they're all
SIP, you can use $DIALSTATUS to continue checking ZAP,MGCP,ETC...
and so on...
-Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple lines with SIP like MGCP?
We have a Dlink DVG-1120M and were surprised that it was able to handle 2 simultaneous conversations to 2 seperate phones using only 1 MAC address and 1 IP address. So we asked ourselves..why can't other 1 MAC/1IP devices do this as well? I have a Grandstream 486 that has 1IP and 1MAC. But I don't see anywhere in sip.conf to add a second line to a device. Is this possible? Can this only be done with an MGCP device? Thanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom Programmable button Mini Howto and ringstate patch
It's very possible that the Polycom IP600 will work with this. As it is just an implementation of a SIP standard for subscribing to the state of other extensions. As for the feature improvements you might see them from me, but not very likely. It is easier for me to train my customers to hit *8 (I will probably just program a pickup button for them) than it is for me to figure out what I have to do in code to accomplish a call pickup. The conference stuff already works satisfactorily. If a person is on the phone you see their button lit, if you hit the button it calls them. They hit ok to accept your call and their existing call goes on hold. If they wish to conference they can this hit their conference button to bridge the three of you together. This is purely a function of the phone. More complex conferences I will achieve with use of the conference application and the flash control panel. You might, however, see the call parking bounty fulfilled by me when I get the time. David Hinkle -Original Message- From: John Todd [mailto:[EMAIL PROTECTED] Sent: Monday, August 30, 2004 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Snom Programmable button Mini Howto and ringstate patch At 1:23 PM -0500 on 8/30/04, David Hinkle wrote: The snom 200 and 220 have five programmable buttons. Each button has a led that can be used to indecate if an extension is idle, in use, or ringing. A button pannel for the 220 is also comming out soon that will have 20'ish programmable buttons on board. This is a killer app for any company that has receptionists handle calls, and pretty usefull for everyone else. As a matter of fact, Asterisk already supports phone idle/in use states for the buttons, and at the bottom of this message you will find a patch to enable the ring state. Howto: 1. Configure the programable buttons as destination and enter the extension in the field. After saving the page the phone will convert the extension to a sip url, which is fine. 2. Modify your asterisk dialplan to provide hints that map a given extension to a channel. (In asterisk, a channel can be busy or ringing, but an extension is just a string of numbers that activate one or more applications). Asterisk seems to provide syntax for allowing more than one channel to be mapped to any particular extension with the hint system, but I did not investigate that. Example: exten = 200,hint,SIP/RonC exten = 200,1,Macro(stdexten,SIP/RonC) exten = 201,hint,SIP/JeanK exten = 201,1,Macro(stdexten,SIP/JeanK) exten = 202,hint,SIP/JeffT exten = 202,1,Macro(stdexten,SIP/JeffT) 3. You must reload the dialplan and then reboot the phone for it's subscriptions to take effect. After that, you should have working lights. 4. If you want the lights to blink on ringing, apply the following patch to the asterisk code. You can not pick up a call by hitting the blinking button, I was going to do this work but I decided to just train the receptionists to hit *8 instead. I have not studied this extensivly, but to implement it, i think it would just require asterisk to have support for sip replaces (I don't know if asterisk supports this or not) and the ringing notify needs to go out with a few more fields. (It seems that the snom phone contacts the sip device listed in one of the ring notify message fields with an invite including a replaces header to pick up a call) I have also included a sip trace of a snom phone picking up a call placed to another phone using the blinking button in case anybody out there wants to tackle this problem themselves (Sample trace was collected when using snom phones with snom's sip proxy software). Please note that it seems like we must include the extra fields in the ring notify before the snom phone will procude the proper replaces invite in order to do a standards compliant call pickup. Notes on patch: If this patch is not in the proper format for submissions please provide me a link to the asterisk submission policies. It has been tested here at DerbyTech for about a week on our live phone system. I submit this patch to the asterisk project under the GPL with hope that it will be resubmited to CVS. Thankyou, David Hinkle Sr. Linux Engineer DerbyTech This is pretty cool! I might get a Snom phone just to try them out. You asked for comments, so here are a few: 1) Send the patch in diff -u format; that's the format used in the bugtracker. 2) You'll need to sign the disclaimer on the http://bugs.digium.com/ interface. This disclaimer doesn't have much of a downside, and all patches to Asterisk for the public CVS have to be disclaimed in this way (avoids SCO-type lawsuits, etc.) 3) Have you looked at the configuration options for the Polycom IP600 phones? I don't know if this trick works with them, but they are pretty slick and have very programmable interfaces which may be almost compatible (or completely compatible) with
Re: [Asterisk-Users] multiple lines with SIP like MGCP?
The HT486 is a single-line device with a PSTN pass-thru. The only multiline IADs I know of are the SIPURAs and the Cisco ATA-186... What you do is you create 2 contexts, 1 for each line of the device and you set the host name to the IP address (or host name if applicable) of the IAD. Set the username of each context to the line's respective extension in Asterisk. Then in the web setup for the IAD, there should be a place to put the username for each line as well as the password... I have not tried this but it should work, SIP is not IP/MAC based it's more like SMTP, it's user based... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog lines and TDM card
Hi, sorry to bother you, but i need to connect 8 standard analog lines to 2 asterisk servers (one in Italy (4 lines) and one in USA (4 lines)) and after let this 2 systems to interact between them. I was thinking to use the TDM400 card equipped with 4 FXO modules on both sides. Is it correct to do this (use the TDM card to terminate analog lines) or i have to use 4 X100P PCI card in both servers? Thank you in advance. Bye, Marcello ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error: CDR on channel 'unknown' has not started
Hi, I installed asterisk-addons and configured it so that the cdr is done on a mysql database. Everything was fine, until I originated outgoing calls using the manager API. The call itself is performed perfectly, but when I hangup, I get the following warning on asterisk CLI: Aug 31 14:29:23 WARNING[-308995152]: cdr.c:331 ast_cdr_end: CDR on channel 'unknown' has not started Aug 31 14:29:23 WARNING[-308995152]: cdr.c:331 ast_cdr_end: CDR on channel 'unknown' has not started Aug 31 14:29:23 WARNING[-308995152]: cdr.c:478 ast_cdr_post: CDR on channel 'unknown' lacks start Aug 31 14:29:23 WARNING[-308995152]: cdr.c:118 ast_cdr_free: CDR on channel 'unknown' lacks start And the cdr record about this call is this: | | | | | | | | |||| 1969-12-31 21:00:00 | 1093973363 | 0 | UNKNOWN |0 | the date/time is always 1969-12-31/21:00:00. What could be wrong? I appreciate any help... Thank you very much ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple lines with SIP like MGCP?
We have a Dlink DVG-1120M and were surprised that it was able to handle 2 simultaneous conversations to 2 seperate phones using only 1 MAC address and 1 IP address. So we asked ourselves..why can't other 1 MAC/1IP devices do this as well? I have a Grandstream 486 that has 1IP and 1MAC. But I don't see anywhere in sip.conf to add a second line to a device. Is this possible? Can this only be done with an MGCP device? I don't have a Granstream, but the Cisco and Snom does that. There are no standards that dictate an IP per line. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can asterisk detect BUSY signal?
Hi, suppose I have agents waiting on a queue and I configure asterisk to dial out and to forward the call to the first agent enqueued. Asterisk will do it even if the answer to the call is busy. Is it possible to configure asterisk to detect the busy signal and, in that case, dial another number, without wasting agent's time? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Client
Hello all, I'm working an a switchboard console for Asterisk and would like to investigate using IAX Client library to Asterisk. I don't seem to be able to find the source. I'm planning on a Win32 app. Guidance on where the source isor who to "take" to is requested. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limit the length of extensions
All of my phones use sip, their accounts are in the sip.conf file and
they have the context of 'company' or whatever. These phones need to be
able to call each others extension, as well as dial outside to the real
world. So in that context I put the outbound rules so that the phones
can call out to the pstn, and I put the extensions of all the other
phones in that context so that the phones can call each other.
Different companies wanted it different. ie some wanted just local, or
local and national, or local national and international. Some wanted to
dial 9 to get an outside line, others wanted to be able to dial without
the 9. So with the variance, I chose to put customized outbound
extensions per context.
Chris Shaw wrote:
Why are you including your outbound context into your incoming context in
the first place? That doesn't make any sense?
I'm guessing that because you're using a number in your exten = you're
using an IP channel like SIP or H323? Is this correct? If you're using a
T1/PRI or POTS lines you need to use 's'.
Using your example, your dialplan should look something like this...
[incoming]
exten = 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
exten = 9543340726,2,setcidname(Blocked)
exten = 9543340726,3,setcidnum(00)
exten = 9543340726,4,Goto(companyname,beginmenu,1)
[companyname]
exten = beginmenu,1,SetVar(CALLEDNAME=CompanyName)
exten = beginmenu,2,Wait,1
exten = beginmenu,3,Answer() ; Answer the channel!
exten = beginmenu,4,Background(company-main)
exten = beginmenu,5,Background(ifyouknow)
exten = beginmenu,6,Goto(company_mainmenu,s,1)
exten = 502,1,Dial(SIP/whoever1SIP/whoever2sip/whoever3,30,m)
exten = 507,1,Dial(SIP/daveSIP/jimSIP/lisa,30,m)
...
[company_mainmenu]
exten = s,1,Background(company-nav1)
exten = 1,1,Goto(company_sales,s,1) ; Sales
exten = 2,1,Goto(companyname,502,1) ; Accounting
exten = 3,1,Goto(companyname,508,1) ; Customer Care
exten = 4,1,Goto(companyname,507,1) ; Technical Support
exten = 5,1,Goto(companyname,202,1) ; Human Resources
exten = 6,1,Goto(companyname,202,1) ; Provisioning
exten = 7,1,Goto(companyname,214,1) ; Marketing
exten = 0,1,Goto(companyname,210,1) ; Operator
...
Instead of jumping back and forth like this, I'd use macros to try and
condense the dialplan a bit...
I can help you more with this if you'd like...
Then for people inside the company there's this...
[outbound-local]
exten = _9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,t) ; for
7-digit dialing
exten = _91800NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _91888NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _91877NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _91866NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
[outbound-ld]
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
[outbound-international]
exten = _9011.,Dial(SIP${EXTEN:[EMAIL PROTECTED],60,T)
[office]
include = outbound-local
include = outbound-ld
include = outbound-international
exten = _[1-5]XX,1,Dial(SIP/${EXTEN},25,tT) ; This is assuming they're all
SIP, you can use $DIALSTATUS to continue checking ZAP,MGCP,ETC...
and so on...
-Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] detect telco voicemail stutter-tone
AFAIK, this is not possible - but I'll throw it out there anyhow... I subscribe to telco voicemail, for the event that all my pstn lines are in use. Telco gives me a stutter-tone dialtone when I have a message waiting. Can a Zap card detect this stutter-tone and perform some action? I'm using TDM400P+FXOs and SIP devices. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can asterisk detect BUSY signal?
[EMAIL PROTECTED] wrote: Hi, suppose I have agents waiting on a queue and I configure asterisk to dial out and to forward the call to the first agent enqueued. Asterisk will do it even if the answer to the call is busy. Is it possible to configure asterisk to detect the busy signal and, in that case, dial another number, without wasting agent's time? Are you asking a is this how it works question, or have you tried using queue's and are not getting the intended results? It should be fairly easy to test, and determine what asterisk's response is. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can asterisk detect BUSY signal?
Nope, Asterisk will not do this, at least not without some serious busy-detect action going on and some tinkering with the dial and agents code, in which case any call that is not busy will have to wait a second or two for Asterisk to say that it isn't busy. Another way to go is to look into what the shady-dial people have done with agents/queues, they have probably already figured that part out. Or, you can try a different Asterisk-based predictive dialer: VICIDIAL, which is a part of the astGUIclient suite: http://astguiclient.sf.net/ It's got web-based management, a cross-platform GUI client, it'll run across multiple Asterisk servers and it's also free. MATT--- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 2:12 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Can asterisk detect BUSY signal? Hi, suppose I have agents waiting on a queue and I configure asterisk to dial out and to forward the call to the first agent enqueued. Asterisk will do it even if the answer to the call is busy. Is it possible to configure asterisk to detect the busy signal and, in that case, dial another number, without wasting agent's time? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Losing voice on Digium demo server - how to spot problem ?
Hi, I'm trying to get Asterisk working on P4 2.8 server behind NAT and Firewall (all ports we're set according to instructions) on DSL line. When pbx connects to Digium demo server( I'm located in Slovenia, Europe), it gets first few words, then silence and then comes back when enumerating dial possibilities (4 for accounting ...). Same happens from SIP or IAX local extension. I guess this is network problem, but would kindly ask for guidance for what measures should I take and what seetings are first to try to avoid this problems. I have another server running at my home on dialup line (28.8kbps) and it connects to digium without problems, so I'm little suspicious being only network traffic problem. Thanks in advance for your effort, regards, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limit the length of extensions
On Wed, 2004-09-01 at 04:39, Deon Rodden wrote:
All of my phones use sip, their accounts are in the sip.conf file and
they have the context of 'company' or whatever. These phones need to be
able to call each others extension, as well as dial outside to the real
world. So in that context I put the outbound rules so that the phones
can call out to the pstn, and I put the extensions of all the other
phones in that context so that the phones can call each other.
Different companies wanted it different. ie some wanted just local, or
local and national, or local national and international. Some wanted to
dial 9 to get an outside line, others wanted to be able to dial without
the 9. So with the variance, I chose to put customized outbound
extensions per context.You should *really* read the examples on the wiki and
asterisk docs, and
other places, but, basically what you should do is this:
in sip.conf all your users belong to the context inside-local or inside-ld or
whatever.
[inside]
exten = 800,1,Dial...
exten = 801,1,Dial...
etc, or use a macro, or whatever you like
[remote]
include = inside
exten = s,1,PlayBack(menu)
etc
[dialout-local]
exten = _9XX,1,Dial(Zap/g1/${EXTEN:1})
exten = _XX,1,Dial(Zap/g1/${EXTEN})
[dialout-ld]
exten = _91NXXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _1NXXX,1,Dial(Zap/g1/${EXTEN})
[inside-local]
include = inside
include = dialout-local
[inside-ld]
include = inside
include = dialout-local
include = dialout-ld
etc...
You should probably have another context for your internal applications
such as voicemail etc, another for global apps (maybe voicemail again,
or meetme, etc) and don't forget to include parking where appropriate.
Asterisk is powerful, and easy to make secure, however, like a lot of
other powerful devices, it is also easy to shoot yourself in the foot.
Regards,
Adam
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Going to voicemail instead of queue if no agent is logged in ?
Hi, I'd like to implement scenario to send user to operator's queue by default (if doesn't dial any extension) but only if there is operator agent logged, so user could get response. If not I'd like to send it to voicemail... Any quick advice ? Thanks in advance, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Headset for Cisco 7960?
Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Headset for Cisco 7960?
I don't know that Plantronics stuff qualifies as inexpensive but I have been using Plantronics H headsets with the adapter at this link. http://store.yahoo.com/founderstelecom/dirconcabfor.html I have two of these cables and they work very well. Bryan Nate Carlson wrote: Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Streaming an audio file to a Zap channel before answer
Hi there
Background:
I want to add DDI and voicemail to users on an existing analogue pabx..
It does not support ISDN.
I have 10 DDI numbers via IAX which I am having sent to my Asterisk
box. I have 2 X100P cards connected to 2 analogue extension ports of my
main legacy analogue pabx. I have set up voicemail for each of my DDI
numbers, and when a call comes in for the person at pabx extension 21, I
do the following:
exten = 21,1,Macro(stdexten,21,Zap/g1c/21)
The c in the Dial command for Standard Extension causes the Zap channel
to not return answerbackto the calling party until the user presses a
'#' key to confirm answer. This is essential because in an
analogue-to-analogue call the only confirmation of answer is tones. I
don't want to use tone detection as it is too unrelaible and the UK
progress tones don't work well with callpogress detection anyway.
In my std-extension macro I include the Dial options r, to allow the
calling party to hear PSTN ringback until the channel is answered,
wither by the called party pressing # or the call going to voicemail.
exten = s,1,Dial(${ARG2},30,tTr); Ring the
interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1)
Everything works as expected, but there is one thing missing. The
called person picks up the phone and hears silence until they press the
# key to answer. This will really confuse my users. I therefore want
to play a helpful message _before the called person confirms answer,
along the lines of 'you have an incoming call. Press the hash key to
accept or hangup.' and loop this until either the person presses the #
key to accept the call, or the dial command times out and the call goes
to voicemail.
I have tried to work around this by using a Perl script in AGI, but the
AGI scripts seem to be single threaded, and exec Background... waits
til the background message has finished before moving on, defeating me.
Anyone got any ideas on this? Anyone hit a similar issue? Any
solutions out there?
Many thanks
Tim Robinson
Basingstoke UK
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Headset for Cisco 7960?
I found the same with lots of headsets and my 7940, but I've just plugged the headset from my Norstar system into the *handset* port on my and it works perfectly. It's not ideal but it'll do for now! Cheers, Benjamin Nate Carlson wrote: I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Headset for Cisco 7960?
Cisco headset pinout is different from normal ones (grr) If it's just for you, (ie nothing too professional ;) you can snip the lead of an existing plantronics type headset and do some reordering - this will give you the necessary info (sorry - can't remember exactly how I did it): http://www.mml.uni-hannover.de/einhorn/headset/index_e.html If you're after something more professional then obviously one of the leads/adapters will be a better approach. HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nate Carlson Sent: 31 August 2004 21:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: Headset for Cisco 7960? Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P No D-channels
Hi Last week I installed Asterisk (release1) with digium t100p single span T1 (wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is currently the only user of the t1 line. All worked well for about a half a day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I consistently get multitudes of blue alarms on all b-channels followed by a loss of d-channel: Aug 31 16:33:49 WARNING[98316]: chan_zap.c:5286 handle_init_event: Detected alarm on channel 1: Blue Alarm Aug 31 16:33:49 WARNING[98316]: chan_zap.c:5286 handle_init_event: Detected alarm on channel 2: Blue Alarm ...etc, intermixed with Aug 31 16:33:49 NOTICE[90123]: chan_zap.c:6920 pri_dchannel: PRI got event: 4 on Primary D-channel of span 1 Aug 31 16:33:49 WARNING[90123]: chan_zap.c:1899 pri_find_dchan: No D-channels available! Using Primary on channel anyway 24! ...and back to reset Aug 31 16:33:54 NOTICE[98316]: chan_zap.c:5281 handle_init_event: Alarm cleared on channel 1... ... Aug 31 16:33:54 NOTICE[90123]: chan_zap.c:6920 pri_dchannel: PRI got event: 5 on Primary D-channel of span 1 ... I found a few hits on VoIP.org and asterisk user forums usually mentioning PCI/BIOS IRQ sharing/conflict, but although I certainly see IRQ misses in zttool as well as /proc/zaptel/1, I cannot see any conflicts - zttool shows blue alarm, recovery and increasing IRQ misses right after zaptel/wct1xxp modprobe and ztcfg. During this search-for-the-truth I disabled all legacy devices (IRQs) I dared, including USB, but to no avail. On Dell GX270, BIOS does not seem to present the option of PCI IRQ line sharing/selection - just a disable/enable option. Mitel 3300 CU (part of 3300 IP-PBX) is set as pri_CPE and * t100p is pri_NET, using esf framing and b8zs code. Wildcard T100P shows green light, our 3300 Mitel CU light on the port I use ranges from yellow (during event recovery) to green (cleared). The telco rep sees nothing wrong with the Mitel - but did reset it several times since this problem started to happen, just to appease me. Zaptel.conf sets t100p to be the primary sync source for the only span, as suggested by many Asterisk users. No changes to Asterisk/Zaptel code has been done since the initial build from the Rel1 FTP site. After spending several days searching on internet, I found a lot of discussion about Digium PRI support which was not totally encouraging. However I am certain it is something simple since I am totally new to Asterisk environment and suspect I am missing something somewhere :( I would welcome any suggestions you may have. Thanks in advance Regards Josef [EMAIL PROTECTED] proc]# cat interrupts CPU0 0:6640846 XT-PIC timer 1:196 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 9: 43296 XT-PIC eth0 10: 12708545 XT-PIC t1xxp 12: 1422 XT-PIC PS/2 Mouse 14: 84025 XT-PIC ide0 15: 256596 XT-PIC ide1 NMI: 0 ERR: 1 [EMAIL PROTECTED] proc]# cat pci PCI devices found: Bus 0, device 0, function 0: Host bridge: PCI device 8086:2570 (Intel Corp.) (rev 2). Prefetchable 32 bit memory at 0xe800 [0xefff]. Bus 0, device 1, function 0: PCI bridge: PCI device 8086:2571 (Intel Corp.) (rev 2). Master Capable. Latency=64. Min Gnt=8. Bus 0, device 30, function 0: PCI bridge: Intel Corp. 82801BA/CA/DB PCI Bridge (rev 194). Master Capable. No bursts. Min Gnt=2. Bus 0, device 31, function 0: ISA bridge: PCI device 8086:24d0 (Intel Corp.) (rev 2). Bus 0, device 31, function 1: IDE interface: PCI device 8086:24db (Intel Corp.) (rev 2). IRQ 9. I/O at 0x1f0 [0x1f7]. I/O at 0x3f6 [0x3f6]. I/O at 0x170 [0x177]. I/O at 0x376 [0x376]. I/O at 0xffa0 [0xffaf]. Non-prefetchable 32 bit memory at 0xfebffc00 [0xfebf]. Bus 0, device 31, function 2: IDE interface: PCI device 8086:24d1 (Intel Corp.) (rev 2). IRQ 9. I/O at 0xfe00 [0xfe07]. I/O at 0xfe10 [0xfe13]. I/O at 0xfe20 [0xfe27]. I/O at 0xfe30 [0xfe33]. I/O at 0xfea0 [0xfeaf]. Bus 0, device 31, function 3: SMBus: PCI device 8086:24d3 (Intel Corp.) (rev 2). IRQ 5. I/O at 0xefe0 [0xefff]. Bus 1, device 0, function 0: VGA compatible controller: PCI device 10de:0181 (nVidia Corporation) (rev 162). IRQ 11. Master Capable. Latency=64. Min Gnt=5.Max Lat=1. Non-prefetchable 32 bit memory at 0xfd00 [0xfdff]. Prefetchable 32 bit memory at 0xf000 [0xf7ff]. Bus 2, device 10, function 0: Network controller: Tiger Jet Network Inc. Model 300 128k (rev 0). IRQ 10. Master
RE: [Asterisk-Users] OT: Headset for Cisco 7960?
GN-Netcom has a nice little headset for about US $120. As to the pin-out, I believe that the headset port uses pins 14 instead of 23. Dan -Original Message- From: Edward Eastman [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 1:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OT: Headset for Cisco 7960? Cisco headset pinout is different from normal ones (grr) If it's just for you, (ie nothing too professional ;) you can snip the lead of an existing plantronics type headset and do some reordering - this will give you the necessary info (sorry - can't remember exactly how I did it): http://www.mml.uni-hannover.de/einhorn/headset/index_e.html If you're after something more professional then obviously one of the leads/adapters will be a better approach. HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nate Carlson Sent: 31 August 2004 21:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: Headset for Cisco 7960? Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Headset for Cisco 7960?
Started a Wiki page here: http://www.voip-info.org/wiki-Cisco+Phone+Headsets Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Edward Eastman To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, August 31, 2004 10:28 AM Subject: RE: [Asterisk-Users] OT: Headset for Cisco 7960? Cisco headset pinout is different from normal ones (grr)If it's just for you, (ie nothing too professional ;) you can snip the leadof an existing plantronics type headset and do some reordering - this willgive you the necessary info (sorry - can't remember exactly how I did it):http://www.mml.uni-hannover.de/einhorn/headset/index_e.htmlIf you're after something more professional then obviously one of theleads/adapters will be a better approach.HTHEd-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of Nate CarlsonSent: 31 August 2004 21:05To: [EMAIL PROTECTED]Subject: [Asterisk-Users] OT: Headset for Cisco 7960?Sorry, I know it's OT, but does anyone know of a relatively inexpensiveheadset that is compatible with the Cisco 7960?I've tried the headset off Norstar phones, doesn't seem to work with orwithout the amp.| nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com || depriving some poor village of its idiot since 1981 |___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P No D-channels
Zaptel.conf sets t100p to be the primary sync source for the only span, as suggested by many Asterisk users. I'm trying to understand so please bear with me... The T100P is connected directly to the Mitel? Or to the Telco through a T1? What I mean is are calls coming into the Mitel from the telco and then from there going into * or are calls going into * first and then being fed into the Mitel? If your T100P is connected to the telco then the clocking source should be the telco as their clocks are going to be a LOT more accurate than your PC's interrupt timers... If your T100P is connected to the Mitel, then you've got it right... Just checking, I wasn't sure from your description... Occasional interrupt misses are pretty normal although in a perfect world with a good mobo they should not happen at all... If you're seeing multiple misses per second (e.g. everytime you do cat /proc/interrupts you see more) then there's a problem... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Headset for Cisco 7960?
I use a Plantronics Supra H51 plugged straight into the headset port, and it works great. B. J. -Original Message- From: Nate Carlson [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 15:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: Headset for Cisco 7960? Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P No D-channels
Hi Chris, thanks for taking time to look this over. T100P/* is connected to the Mitel IP-PBX/CU and it to telco - so I think our setting is correct. BTW, I did try 0 (as well as 2) without success, just for fun, before I came on a good explanation of the sync source in this forum. Unfortunately, I am seeing great many missed IRQs continually...if in fact it is that which causes the loss of D-channel. Regards Josef -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: August 31, 2004 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T100P No D-channels Zaptel.conf sets t100p to be the primary sync source for the only span, as suggested by many Asterisk users. I'm trying to understand so please bear with me... The T100P is connected directly to the Mitel? Or to the Telco through a T1? What I mean is are calls coming into the Mitel from the telco and then from there going into * or are calls going into * first and then being fed into the Mitel? If your T100P is connected to the telco then the clocking source should be the telco as their clocks are going to be a LOT more accurate than your PC's interrupt timers... If your T100P is connected to the Mitel, then you've got it right... Just checking, I wasn't sure from your description... Occasional interrupt misses are pretty normal although in a perfect world with a good mobo they should not happen at all... If you're seeing multiple misses per second (e.g. everytime you do cat /proc/interrupts you see more) then there's a problem... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware suggestion
Hi, Can anyone recommend a BRI card which works fine with asterisk and which supports point-to-point mode? Software fax detection should also work. Price does not matter. :) Digium seems to sell only PRI cards, and the Beronet drivers for the quad BRI cards seem to be in an early stage of development (besides, fax detection seems not to be implemented). Thanks pm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why is it called 'Comedian Mail?
Inquiring (management) minds want to know. I'm assuming it's because 'it's funny how simple it really is to write a really decent voicemail system'? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
