Re: [asterisk-users] [SOLVED]Linksys SPA 3102 causing me problems
On Thursday 29 March 2007 23:26, Alan Chandler wrote: I have a linksys SPA 3102 with a DECT phone connected into its Telephone port. It has been working, but something I've done (and I don't know what) means that now everytime asterisk tries to dial it, it says it is busy. In the end I did a factory reset and then reloaded all my parameters, and now it works great. -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on Priorities
Hi, I am attempting to change my dialplan to use 'n' priorities and labels for easier reading, and less re-numbering :) but how do you handle the plus 101 ? In my extensions.conf I have a simple plan for testing :- [inbound-sip] exten = uxbod,1,Dial(sip/1001,20,t) exten = uxbod,n,PlayBack(uxbod) exten = uxbod,n,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,n,Hangup() exten = uxbod,103,PlayBack(uxbod) exten = uxbod,104,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,105,Hangup() So when the extension has to add 101 do I just do n+101 ? TIA -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on Priorities
[inbound-sip] exten = uxbod,1,Dial(sip/1001,20,jt) exten = uxbod,n,Hangup exten = uxbod,102,PlayBack(uxbod) exten = uxbod,103,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,104,Hangup() here if dial fails then n+101 =102 extension will get executed unless you use j option in dial application and priority jumping has to be set to priorityjumping=yes in the general section of your extensions.conf file. In your dialplan i dont know y you r forcing the caller to goto voicemail even if the call has already answered. I hope you understand my modification in your dialplan. On 3/31/07, --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Hi, I am attempting to change my dialplan to use 'n' priorities and labels for easier reading, and less re-numbering :) but how do you handle the plus 101 ? In my extensions.conf I have a simple plan for testing :- [inbound-sip] exten = uxbod,1,Dial(sip/1001,20,t) exten = uxbod,n,PlayBack(uxbod) exten = uxbod,n,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,n,Hangup() exten = uxbod,103,PlayBack(uxbod) exten = uxbod,104,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,105,Hangup() So when the extension has to add 101 do I just do n+101 ? TIA -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on Priorities
also only priorities are added incase of priority jumping, not extensions. On 3/31/07, --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Hi, I am attempting to change my dialplan to use 'n' priorities and labels for easier reading, and less re-numbering :) but how do you handle the plus 101 ? In my extensions.conf I have a simple plan for testing :- [inbound-sip] exten = uxbod,1,Dial(sip/1001,20,t) exten = uxbod,n,PlayBack(uxbod) exten = uxbod,n,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,n,Hangup() exten = uxbod,103,PlayBack(uxbod) exten = uxbod,104,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,105,Hangup() So when the extension has to add 101 do I just do n+101 ? TIA -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme question
Hi, I'm experimenting with the Meetme feature of Asterisk 1.2, exten = 2095,1,MeetMe(|Ds) This almost gives me what I want, where each employee can create their own on-the-fly conferences with a personal Conference Number and PIN. However, as the PIN is actually set by the first callee, then its subject to problems (first callee might enter the wrong PIN, and then no-one else can join). What I really want is something that covers the below: - One call-in number - Employees get their own unique conference # (this could be their own extension), and can set a public PIN that only they can change. - I don't really want a www-based system, as most of my users are usually mobile, and might not have access to the corporate intranet. Thanks, Adrian Marsh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wireless desktop phones
On 28 Mar 2007, at 21:51, Matt Gorecki wrote: I'm also in the market for a wi-fi phone. My boss currently has a cordless phone and wants to keep the same functionality. We have a robust wireless network in the office and the phone will be staying here, so roaming is not really an issue. Everybody in the office is still going to get wired phones regardless. I got a couple of nokia e60's and despite being a _royal_pain_ to configure I'm pretty happy with them. Don't give one to anyone who can't program their own VCR, the interface is a bit daunting at first. It's a delight to have your cell phone be your officephone the moment you step into the wifi pool :-) Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: wireless desktop phones
JN == Jordan Novak [EMAIL PROTECTED] writes:
JN Okay, I get it. I still have a problem though. I have no way to
JN wire 30% of these end-points. P{hysically impossible. They do have
JN cat3 twisted pair to each phone. But of course they want IP. Are
JN there any adpaters that will give me just enough bandwidth to get
JN it done. The computer network is all wireless so the phones would
JN have all the bandwidth.
HomePNA should do what you want.
/Benny
___
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Re: [asterisk-users] Multi-Level Queue
Peder @ NetworkOblivion wrote: I also had a question about acking a call. It appears that acking a call is under agents.conf. I want to specify members as SIP/1234, etc, rather than having users login all the time. I don't want to have to login from my cell, I would prefer it to just know that my cell number is always a member. Is it possible to force an ack of a call if I define members as SIP/? Not directly, no, because channel drivers don't implement call acking (except for chan_agent). However, if you create a context in your dialplan that uses Dial() to call the SIP device with acking turned on, then you can add Local/[EMAIL PROTECTED] as a member of the queue and have those calls run through the Dial() application. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Level Queue
On 3/31/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: Peder @ NetworkOblivion wrote: I also had a question about acking a call. It appears that acking a call is under agents.conf. I want to specify members as SIP/1234, etc, rather than having users login all the time. I don't want to have to login from my cell, I would prefer it to just know that my cell number is always a member. Is it possible to force an ack of a call if I define members as SIP/? Not directly, no, because channel drivers don't implement call acking (except for chan_agent). However, if you create a context in your dialplan that uses Dial() to call the SIP device with acking turned on, then you can add Local/[EMAIL PROTECTED] as a member of the queue and have those calls run through the Dial() application. ___ Or he could use app_followme which has call acking built in. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem while using asterisk Realtime
- Sanjay Rajdev [EMAIL PROTECTED] wrote: Another thing I found is if i do the following # make menuselect go in option 2. Call Detail Recording here the option 4. cdr_odbc and option 5. cdr_pgsql both have XXX marked infront of them. And at the bottom of screen it says ODBC CDR Backend Depends on: unixodbc(E) I donot know why it says so as I have already mentioned below that the odbc connectivity is working fine. Also I have checked other option in Menuselect everywhere it says same for odbc. Can someone please let me know what I is wrong here. You probably need to install libtool. After installing it, re-run the configure script. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sponsored development - Monodirectional audio handling
Hi Guys, we're needing a special implementation on Asterisk Our intention is to contribute the development and share back the code to Asterisk community Here is what we need: - An option to Asterisk Dial command which, if used, when calls is answered gives monodirectional audio (Caller should hear the called party but not vice-versa) - A DTMF sequence (maybe handled in features.conf) for the Caller to start to have bidirectional audio - When the Callers makes the audio 'bidirectional' an Event should be generated so that we can see it from the manager API The purpose of thisi implementation is to deal with some carriers that give us the call as ANSWERED when the called party is still ringing. Our billing software is billing the user (and the carrier is billing us) even with unsuccessful calls. This way we can start billing when the user press the DTMF sequence to unlock audio (even if carriers bill us wrongly) Someone wants to help ?? Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : RE : [asterisk-users] wireless desktop phones
Hi Tobias and the list, Yes, I have, I use and sell them to integrators ;-) But only the 600v3 family, not the older ISND or analog versions, and the current DECT handsets 40XX. Any Digium interfaces run well with them as any SIP IP-Phone, of course. The sound quality is GREAT and the infrastructure deployment possibilities wonderfull and scalable ! But, you must run a training with the company to well understand the how to do and capture the knowledge. I must also say that I am a radio guru and it's certainly easyer for me to understand this kind of equipments and how to avoid the deployment traps that an engineer who doesn't know what are radiocommunications but only VoIP. I have run them behind all current Asterisk versions, including the ASteriskNOW. Check about your codecs as usual. The last firmware from this last week suppresses few minor buggs occured during roaming in few previous cases. Best Regards, Francois BERGERET, France. -Message d'origine- De : Tobias Wolf [mailto:[EMAIL PROTECTED] Envoyé : jeudi 29 mars 2007 16:23 À : [EMAIL PROTECTED] Objet : Re: RE : [asterisk-users] wireless desktop phones [EMAIL PROTECTED] schrieb: Hi the list, Think Kirk solution ;-) www.kirktelecom.com Do you have this working in you enviroment ? Currently I have some test devices from Kirk (KIRK Wireless Server 600/3 with SIP protocoll and a couple of handsets). But i am not able to get audio between the handset and the destination then i call a zap channel. Calling another Kirk handset or another SIP phone (Snom) works quite well, then i dont put any options in the Dial Command. Otherwise i dont't get any audio also. Signalling a call is no problem. It would be great to hear from you if your setup work perfectly and what your enviroment is (Asterisk Version, type of Kirk Server). Thanks in advance, Tobias Wolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sponsored development - Monodirectional audio handling
You could put a bounty on this. You may find someone who will be willing to write this for money. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edoardo Serra Sent: Saturday, March 31, 2007 11:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sponsored development - Monodirectional audio handling Hi Guys, we're needing a special implementation on Asterisk Our intention is to contribute the development and share back the code to Asterisk community Here is what we need: - An option to Asterisk Dial command which, if used, when calls is answered gives monodirectional audio (Caller should hear the called party but not vice-versa) - A DTMF sequence (maybe handled in features.conf) for the Caller to start to have bidirectional audio - When the Callers makes the audio 'bidirectional' an Event should be generated so that we can see it from the manager API The purpose of thisi implementation is to deal with some carriers that give us the call as ANSWERED when the called party is still ringing. Our billing software is billing the user (and the carrier is billing us) even with unsuccessful calls. This way we can start billing when the user press the DTMF sequence to unlock audio (even if carriers bill us wrongly) Someone wants to help ?? Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: wireless desktop phones
You can always using a gaming bridge for phones that do not support
wireless.
I've done this before with this:
Linksys / WGA54G / 54Mbps / 802.11g / Wireless Bridge
Setup is pretty easy.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benny Amorsen
Sent: Saturday, March 31, 2007 9:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: wireless desktop phones
JN == Jordan Novak [EMAIL PROTECTED] writes:
JN Okay, I get it. I still have a problem though. I have no way to
JN wire 30% of these end-points. P{hysically impossible. They do have
JN cat3 twisted pair to each phone. But of course they want IP. Are
JN there any adpaters that will give me just enough bandwidth to get
JN it done. The computer network is all wireless so the phones would
JN have all the bandwidth.
HomePNA should do what you want.
/Benny
___
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[asterisk-users] LUSYN patches
Hello all, The history buffer patches maintained by myself have been updated to work with Zaptel 1.4.1 and Asterisk 1.4.2. They are available along with installation instructions at www.lusyn.com. Rgds, Marc LUSYN Limited is a company registered in England and Wales with company number 04803242 and VAT number 817 9045 14. The registered office is : 3 Teal House The Millstream London Road High Wycombe Buckinghamshire HP11 1AE LUSYN and the LUSYN logo are registered trademarks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LUSYN patches
On Sat, Mar 31, 2007 at 05:17:02PM +0100, Marc McLaughlin wrote: Hello all, The history buffer patches maintained by myself have been updated to work with Zaptel 1.4.1 and Asterisk 1.4.2. They are available along with installation instructions at www.lusyn.com. BTW: for zaptel 1.2 = 1.2.13 you need to slightly edit the patch and use zaptel-base.c instead of zaptel.c . In asterisk 1.2.15 there are also some minor changes. A new version could be found at http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/patches/ukcid.dpatch?op=filerev=3132sc=0 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Sponsored development - Monodirectional audio handling
Salvatore Giudice ha scritto: You could put a bounty on this. You may find someone who will be willing to write this for money. My Bounty for that feature is 500 USD -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edoardo Serra Sent: Saturday, March 31, 2007 11:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sponsored development - Monodirectional audio handling Hi Guys, we're needing a special implementation on Asterisk Our intention is to contribute the development and share back the code to Asterisk community Here is what we need: - An option to Asterisk Dial command which, if used, when calls is answered gives monodirectional audio (Caller should hear the called party but not vice-versa) - A DTMF sequence (maybe handled in features.conf) for the Caller to start to have bidirectional audio - When the Callers makes the audio 'bidirectional' an Event should be generated so that we can see it from the manager API The purpose of thisi implementation is to deal with some carriers that give us the call as ANSWERED when the called party is still ringing. Our billing software is billing the user (and the carrier is billing us) even with unsuccessful calls. This way we can start billing when the user press the DTMF sequence to unlock audio (even if carriers bill us wrongly) Someone wants to help ?? Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 129
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 129
[EMAIL PROTECTED] wrote: Je suis absent du 2/04/2007 au 11/04/2007. Do not auto-reply to list messages. :) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off Topic: Open Source USB Softphone
sip would be the required one as iax..well.. also openwengo wont work.. to much overhead .. broswrer needed.. ie component + flash + css+js etc.. not viable.. so im also asking anyone have one ? since ihave a supply of around 2000 of the vonage usb stick OEM.. On 3/30/07, Michael Van Donselaar [EMAIL PROTECTED] wrote: Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Understanding the dial flags
I am trying to make a system where a conference user can invite others
to join. I am running the 1.2 version of asterisk, so can't use the
example on voip-info.org.
With use of the X flag on a meetme conference to exit with a single
digit, I can get people to join me in a conference with
exten =
_XXX,1,Dial(${THEIR_EXTEN},,dG(conference-context^${CALLERID}^1))
where the conference-context has a something like this
[conference-context]
exten = _XXX,1,MeetMe(${EXTEN},XMsa)
exten = _XXX,2,MeetMe(${EXTEN},Ms)
The problem with that approach is that you never get to talk to the
called party and ask whether they want to join the conference or not.
So I thought I would take another approach and try change the Dial
statement
exten = _XXX,1,Set(GOTO_ON_BLINDXFER=conference^${CALLERID}^2)
exten = _XXX,2,Dial(${THEIR_EXTEN},,dTg)
exten = _XXX,3,GoTo(conference-context,${EXTEN},1)
Now, when I blind xfer the user he goes straight into the conference.
Unfortunately, the g flag - which should mean I carry on down the dial
plan doesn't appear to work in this case. When he hangs up, I do go
back to the conference, but when he is transfered and goes to the
conference, I seem to end in limbo.
I even tryed to add an h extension to the same context as the Dial is
in, but that didn't seem to help.
Is there a way out of this?
--
Alan Chandler
http://www.chandlerfamily.org.uk
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[asterisk-users] Setting a call to be recorded before Xfer?
I need to allow the company operator to decide whether to record a call. (Car dealership that needs to coach salespeople). They don't want to record every sales call, just for the purposes of coaching certain employees on an ad hoc basis. The situation is: a. Call comes in on PSTN PRI b. Call is routed to operator in dialplan c. Operator ascertains that its a sales call for a salesman in need of coaching d. Operator needs to set the call to record status e. Operator xfers the call to the salesman f. Salesman answers the call, completes the communication and hangs up. g. Sales manager can review the call and coach the salesman. Since a new channel is created at the point of transfer, how can I flag the channel to be created as one to be recorded? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Setting a call to be recorded before Xfer?
Maybe have 2 extensions for all the sales people. Calls to 7xx ring their desk extension Calls to 8xx ring their desk extension but also send mp3 of the call via email to sales person and sales manager. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J French Sent: Saturday, 31 March 2007 6:37 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Setting a call to be recorded before Xfer? I need to allow the company operator to decide whether to record a call. (Car dealership that needs to coach salespeople). They don't want to record every sales call, just for the purposes of coaching certain employees on an ad hoc basis. The situation is: a. Call comes in on PSTN PRI b. Call is routed to operator in dialplan c. Operator ascertains that its a sales call for a salesman in need of coaching d. Operator needs to set the call to record status e. Operator xfers the call to the salesman f. Salesman answers the call, completes the communication and hangs up. g. Sales manager can review the call and coach the salesman. Since a new channel is created at the point of transfer, how can I flag the channel to be created as one to be recorded? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Meetme question
If you know what you want the conf room number to be, then set that up in
meetme.conf.
You would have to write your own IVR though, and use Authenticate() with the
PIN kept in the DB. Its a hack but it would do what you want:
exten = _X,1,Playback(conf-getconfno)
exten = _9XX,1,Authenticate(${DB(conf/${EXTEN})})
exten = _9XX,1,MeetMe(${EXTEN},s,)
I don't know any other way to have a PIN number that isn't statically
defined in the meetme.conf file or created on the fly with the room useing
the D option, if you know another way let me know.
--
Justin
Hi,
I'm experimenting with the Meetme feature of Asterisk 1.2,
exten = 2095,1,MeetMe(|Ds)
This almost gives me what I want, where each employee can create their own
on-the-fly conferences with a personal Conference Number and PIN. However, as
the PIN is actually set by the first callee, then its subject to problems
(first callee might enter the wrong PIN, and then no-one else can join).
What I really want is something that covers the below:
- One call-in number
- Employees get their own unique conference # (this could be their own
extension), and can set a public PIN that only they can change.
- I don't really want a www-based system, as most of my users are usually
mobile, and might not have access to the corporate intranet.
Thanks,
Adrian Marsh
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Re: [asterisk-users] Linksys SPA 3102 causing me problems
On 3/30/07, Gergo Csibra [EMAIL PROTECTED] wrote: Friday, March 30, 2007, 5:02:08 AM, Matt wrote: Wehh... He activated the DND function of Linksys. It can be activate with *78 and deactivate with *79. No because if that was a case his sip trace would show something along the lines of 486 BUSY HERE from the ATA. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Level Queue
So does the P option in app_dial seems to me the easiest way to implement is just hack app_dial so it wont prompt to record the name. On 3/31/07, BJ Weschke [EMAIL PROTECTED] wrote: On 3/31/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: Peder @ NetworkOblivion wrote: I also had a question about acking a call. It appears that acking a call is under agents.conf. I want to specify members as SIP/1234, etc, rather than having users login all the time. I don't want to have to login from my cell, I would prefer it to just know that my cell number is always a member. Is it possible to force an ack of a call if I define members as SIP/? Not directly, no, because channel drivers don't implement call acking (except for chan_agent). However, if you create a context in your dialplan that uses Dial() to call the SIP device with acking turned on, then you can add Local/[EMAIL PROTECTED] as a member of the queue and have those calls run through the Dial() application. ___ Or he could use app_followme which has call acking built in. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP Phones have buttons can be assigned to functions with Asterisk
bilal ghayyad wrote: I heared that polycom needs adaptor for the power as it does not provide standard PoE, also I do not know this. You need a special cable to use a 501 with a POE switch, that's all. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Paging
From: Forrest Beck [EMAIL PROTECTED]
Date: Fri, 30 Mar 2007 16:52:39 -0400
Forgot to mention.
We are using Polycom phones on asterisk 1.4.2
I tried the allpage agi, but it checks for all SIP peers connected to
the server.
On 3/30/07, Forrest Beck [EMAIL PROTECTED] wrote:
First off, A lot of thanks to this list. I have learned ton from
reading through the posts this past year.
I need some advise.
I have two group of phones connected to a single server.
Group1= SIP/2503SIP/2504
Group2=SIP/3501SIP/3502
I'd like to be able to dial an extension and page a certain group of
phones only if ChanIsAvail returns 1.
I am not sure how to go about programming this. I though to write a
AGI script that reads a list of phones (one list per group), checks
ChanIsAvail then Pages the phone. I will have about 60 extensions
per group to Page. Will there be lag until all the phones get paged
and the script finishes?
The lag shouldn't be too large. Yet you don't have to use AGI to build a
list, and you don't even have to wait for all channels to be checked if I
understand the objective correctly. For example (untested),
exten = _Z.,1,ChanIsAvail(SIP/${EXTEN},j)
exten = _Z.,n,Dial(SIP/${EXTEN})
exten =
_Z.,101,Set(group=$[$[${GROUP1}=~SIP/${EXTEN}]?${GROUP1}::${GROUP2}])
exten = _Z.,n,While(${group})
exten = _Z.,n(check),ChanIsAvail(${group},j)
exten = _Z.,n,Set(page=${${page}${AVAILORIGCHAN}}); tweak if empty not
acceptable
exten = _Z.,n,Dial(${page}); start dialing before list completes
exten = _Z.,n,Set(group=$[${group}=~${AVAILORIGCHAN}*(.*)])
exten = _Z.,n,Endwhile
exten = _Z.,check+101,Congestion; or however way you want to handle no
channel available
You may need to tweak a bit to get it working but that's the spirit. Hope
this helps.
Yuan Liu
Then I thought maybe a Macro in the dialplan to dial a global var of
the group of phones, but that won't work. If phone isn't available,
none will get paged.
Has anyone done this before? I just don't know where to start.
Thanks
--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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Re: [asterisk-users] Question on Priorities
From: Rizwan Hisham [EMAIL PROTECTED] Date: Sat, 31 Mar 2007 17:01:51 +0500 [inbound-sip] exten = uxbod,1,Dial(sip/1001,20,jt) exten = uxbod,n,Hangup exten = uxbod,102,PlayBack(uxbod) exten = uxbod,103,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,104,Hangup() here if dial fails then n+101 =102 extension will get executed unless you use j option in dial application and priority jumping has to be set to priorityjumping=yes in the general section of your extensions.conf file. In your dialplan i dont know y you r forcing the caller to goto voicemail even if the call has already answered. I hope you understand my modification in your dialplan. On 3/31/07, --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Hi, I am attempting to change my dialplan to use 'n' priorities and labels for easier reading, and less re-numbering :) but how do you handle the plus 101 ? In my extensions.conf I have a simple plan for testing :- There is a simpler way, by using label. [inbound-sip] exten = uxbod(ntest),1,Dial(sip/1001,20,t) exten = uxbod,n,PlayBack(uxbod) exten = uxbod,n,Hangup() exten = uxbod,ntest+101,PlayBack(uxbod) exten = uxbod,n,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,n,Hangup() Yuan Liu [inbound-sip] exten = uxbod,1,Dial(sip/1001,20,t) exten = uxbod,n,PlayBack(uxbod) exten = uxbod,n,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,n,Hangup() exten = uxbod,103,PlayBack(uxbod) exten = uxbod,104,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,105,Hangup() So when the extension has to add 101 do I just do n+101 ? TIA -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP Phones have buttons can be assigned to functions with Asterisk
http://www.voip-info.org/wiki-Polycom+Phones#Newphones quote: WARNING: The IP 30x and IP 50x models do not have on-board Power Over Ethernet chips. Although the phone claims to support 802.3af and the Cisco POE standard (note it says optional), the an additional cable (see part list above) is required on these models. This raises the list price to $215 or $305 when used in a Power over Ethernet environment; if you know you're going to need PoE, buy the part with the PoE cable included (and no wall power brick) to save money. This warning does not apply to the 60x or any future models Just the 30x and 50x, I just setup an office full of 430s with a 3com Baseline 2426 PWR PoE switch. It is a great low cost setup. Justin On 3/31/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote: bilal ghayyad wrote: I heared that polycom needs adaptor for the power as it does not provide standard PoE, also I do not know this. You need a special cable to use a 501 with a POE switch, that's all. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Justin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
