Re: [Asterisk-Users] skype channel
Message: 10 Date: Sun, 15 May 2005 21:41:23 + From: Laurent Lesage [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] skype channel To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi *, I was just going to ask the same question. Does anybody have an information about Skype and Asterisk? Any link? Thanks in advance I've just added a view day's ago some information on it on the wiki. As far as I know there is nothing really working 'yet' but I'm sure since the API is out it' won't take long :-) http://www.voip-info.org/tiki-index.php?page=bounty%20skype Wessel de Roode Laurent Bartek Kania a icrit : -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (GNU/Linux) iD8DBQFCgLlVWYjaxM2wIe4RAuSKAJ9VNMIO2h838Y2yXAFDAQaJOjPa3gCfeokZ Ghsrpa8Gp3pHt5/bUinZKUA= =fUgt -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Message: 11 Date: Sun, 15 May 2005 17:49:41 -0400 From: Paul [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] knopsterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed trixter http://www.0xdecafbad.com wrote: does anyone have knopsterisk for download, I assume that because its GPL the creator of that iso cant restrict spreading it. A friend wanted it to play on a box and the only thing I can find with google is the knopsterisk.com site which wants $10 to get a copy and does not provide (as far as I can tell) any free distribution access which is his/hers/its/them/they/whatever right (being politically correct is hard). If there is some distribution problem with doing this then I would also appreciate hearing why it cant be distroed by 3rd parties. Thanks The website says /*Now with Asterisk Version 1.0! which makes me wonder how many they have sold. Also makes me wonder if the knoppix part is very up to date. They don't mention licensing/copyright anywhere. We can figure that all the software is on the CD is under free licensing but all they have to do is add a single readme file with a restricted license or copyright and you make identical copies of the CD. I would first try contacting them and get those details. You also want to know where the source is because there might be some modifications they made to knoppix packages or the packages they added. I think you would be better off to make a knoppix CD, boot it and get * installed and running. After that read the following and maybe you can create something better to share with the world. http://www.knoppix.net/wiki/Knoppix_Remastering_Howto */ -- Message: 12 Date: Sun, 15 May 2005 15:55:03 -0600 From: Ira Burton [EMAIL PROTECTED] Subject: [Asterisk-Users] 911 Options To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 I am curious if anybody has pointers on the best way to get the 7 digit PSAP number for an area. I am thinking about making a '911' extension that will dial the PSAP number, wait for the PSAP to answer and play a message giving the address of the originating call, and replay the the information every three minutes. I am concerned what may happen if my children try to dial 911 in an emergency but do not yet know our address. How are other people handling this? -- Message: 13 Date: Sun, 15 May 2005 15:15:15 -0700 From: trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] knopsterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Sun, 2005-05-15 at 17:49 -0400, Paul wrote: I think you would be better off to make a knoppix CD, boot
[Asterisk-Users] Re: Dutch SIP or IAX numbers
Message: 1 Date: Sun, 1 May 2005 19:01:24 +0200 From: Michiel van Baak [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dutch SIP or IAX numbers To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On 12:23, Sun 01 May 05, Asterisk wrote: How knows where I can get a Dutchphone number for asterisk? Pilmo is not delivering one for home use. www.voipgate.nl Not using it, but offers IAX2. Wessel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.0 - Release Date: 29-04-05 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to park/transfer a call received from a Queue?
From: Matias G. [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] How to park/transfer a call received from a Queue? To: Asterisk Users Mailing List - Non-Commercial Discussion you haven't include hte part where you make AgentCallBackLogin() the context you enter there is the one where your call will be tried to place when the agent transfers it ie: exten = 11,1,AgentCallbackLogin(|[EMAIL PROTECTED]) will log that agent in a valid extension inside that context. when the agent tries to transfer he will be allowd to transfer to extensions valid in that context... hope this helps. GREAT! This was the trick! I just needed to add include = parkedcalls In the context of [CallCenter] include = parkedcalls Exten = .. All the phone extensions. And now it's parking and transfering as a charm :-)) Thanks Matias and the other hints I received from the list :-) Wessel de Roode -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.4 - Release Date: 27-03-05 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to park/transfer a call received from a Queue?
Hi! I'm trying to transfer a incomming call from a Queue to another extension. I'm receiving a call from a queue with the AgentCallbackLogin. The queu is as following: Queue(sales|t) Which should allow transfers. So as soon as the call is answered I would like to be able to transfer it When the agent presses the # I get the dialtone but as soon as I press any digit Asterisk tells me that that is a wrong extension? Calling between phones and park calls works fine, so the parking application is working ok. I'm only missing something here with the Queue's. Here are my configuration fragments. extensions.conf: [incoming] include = parkedcalls exten = ,1,Answer exten = ,2,Queue(sales|t) features.conf: [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls Queues.conf: [sales] joinempty = yes announce-frequency = 30 announce-holdtime = yes member = Agent/2537 Please help :-) Thanks in advanced, Wessel de Roode -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.3 - Release Date: 25-03-05 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Is anyone using asterisk in a small call
Date: Fri, 4 Mar 2005 17:37:16 -0500 From: John Scully [EMAIL PROTECTED] Subject: [Asterisk-Users] Is anyone using asterisk in a small call center Hello - I have just joined the lists and am considering installing quite a few * systems. I am looking for an IP-PBX with both solid standard features and call-center/ACD features. I have read the documentation and the list archives and did not see any references to real call-center type reporting and queuing. It is there. Look for the Queue's plugin it is default loaded in * Is anyone out there using * in this kind of environment? The features I would be looking for would include: Yes I'm running it for a bussiness and it is wokring fine. My agent's are loggin in and out by them self or the manager is putting them to work :-) Skill set routing Think you mean prioritizing of your agents that is called pennalty under asterisk multiple inbound queues. MM not sure you mean with that but you can connect queu's real time displays The data is there, you need an application who intreprt this data. Look for the different gui's and software that is out there opensource or closed. We are build our own tailor made applicaton for this. tracking of lost calls, wait times etc. Yes that is default available just type Queue show And * will show you the numbers :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAPHFC is back in bristuff 0.2.0-RC7d+
For the unknown. ZAPHFC is a driver that enables the use of a cheap ISDN card to run in TE or NT mode. In other words, to run like a standard ISDN terminal to receive and place calls over a BRI line. The driver also enables to us a hfc card in NT mode which enables it to connect to your own ISDN pdx as if it was your own telecom provider. The driver van be found on the www.junghanns.net Cards that are build around the hfc chipsets can be found here: http://isdn.jolly.de/cards.html Wessel Here are the latest release notes: 0.2.0-RC7f - D-channel up/down messages in BRI_CPE_PTMP mode will only be shown if asterisk is started with at least - - some sample configurations (SAMPLES directory) 0.2.0-RC7e - added m option to chan_zap, this will provide an outgoing channel without echo cancelation, useful for fax and modem, e.g.: exten = 1234,1,zapEC(off) ; disable EC on the incoming channel exten = 1234,2,Dial(ZAP/g1m/1234) ; create channel without EC 0.2.0-RC7d - zaphfc is back, now also sending/receiving complete HDLC frames - zaphfc B channel improvements (please test!) - added app_zapEC to enable/disable echo cancelation from the dialplan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2: Connection rejected
carrier via a PRI, they will dictate what the DID looks like. Some will be the last 4 digits, others will be all 10. (assuming US). They do this, because it would be to difficult to maintain your extension mapping on their side. You purchase a DID. When a call comes in it says, This is the number they were calling, you do your own matching to whatever extension you want. Now, what about the folks who are trying to call other countries, and potentially be called by other DIDs themselves? I'm assuming this sort of thing is very likely. Did you set a username? On some weired reason that is needed in 1.0.5 for IAX to work. Wessel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dutch VOIP-PSTN provider
Message: 1 Date: Sat, 19 Feb 2005 16:20:31 +0100 (CET) From: Remco Barende [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dutch VOIP-PSTN provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed I read a lot about US providers that can terminate a PSTN number for you and offer IAX or SIP connectivity. Does anyone know such a company in The Netherlands ? I read about Unet. Anyone with experience with them ? Any information is welcome. I'm currently one of a closed group of the test users for a dutch one. On this moment I can only tell you that it is coming soon :-) I'll post it here as soon as it is up running. Wessel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vservices.inv of Julian Pawlowski anoyne has the macro-dailer for this?
Hi, I've found the Vertical Service Codes / vservices.inc of Julian in the cache of google. It's an very extended extensions include with all the *21 *67 etc services implemented so it is stored to ODBC or if you replace it to Dbget/put etc. I'm wondering if somebody has the macro/agi for using these extensions once stored in the Asterisk db or ODBC. Or am I missing something? And will it work just as it is under ODBC Wessel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting call forward for Agent's in a Queue
Hi!,
I'm trying to set up a Queue (which works fine now :-)
Sip clients can login in to the Queue with dialing 91 on there phone.
And as soon as there are customers the Queue calls the agents back.
I would like that the queue calls the agents also if it's phone is
call-forwarded.
With agents (sip clients) are added with the following extensions:
exten = 91,1,AddQueueMember(myqueue)
exten = 91,2,Playback(agent-loginok)
exten = 91,3,Hangup
However if I use the following script to enable call-forwarding:
exten = _*21*X.#,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten = _*21*X.#,2,Answer
exten = _*21*X.#,3,Playback(call-fwd-unconditional,skip)
exten = _*21*X.#,4,Hangup
And the following macro for every internal dial command:
[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten=s,1,DBget(temp=CFIM/${ARG1}) ; Get CFIM key, if not existing, goto 102
exten=s,2,Dial(Local/[EMAIL PROTECTED]/n) ; Unconditional forward
exten=s,3,Dial(${ARG2},20) ; 20sec timeout
exten=s,4,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing, goto
105
exten=s,5,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on busy or unavailable
exten=s,102,Goto(s,3)
exten=s,105,Busy
It is not working as the Queue is dialing directly the extension of the sip
phone.
Any alternatives or workarounds?
Many thanks in advance...
Wessel
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RE: [Asterisk-Users] Eyebeam - asterisk - Messenger
Just add a line to your sip.conf: [general] videosupport=yes And to enable video with eyeBeam press the switchon button on the screen :-) Wessel -Original Message- From: Ing. Ignacio Ortega A. [mailto:[EMAIL PROTECTED] Sent: Friday, January 28, 2005 19:33 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Eyebeam - asterisk - Messenger did you find how to configure video with eyebeam using asterisk because i wasn`t able to do it yet as well i want to se messangin with it ThanK You On Fri, 28 Jan 2005 13:23:46 -0500 (EST), Francois Meehan [EMAIL PROTECTED] wrote: Hi all, I would like to connect in sip mode an Eyebeam client to a messenger via Asterisk. I want to use video. Nat is not an issue as vpn connections will be used. Is this a difficult tasks, can someone give me some pointers to get started... Have a good week-end, Francois Random Thought: --- Wanna buy a duck? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
