Re: [asterisk-users] How to notify an event to every user
2008/9/21 [EMAIL PROTECTED] Hi Olivier, What type of handsets are you using in-house? Hi, I'm using this one http://www.voip-info.org/wiki/view/Thomson+ST2030 I'm not familiar with its paging functions but I think it's time to study them ... (from memory, it should be possible to specify with ALERT-INFO that a call is to be answered automatically in handfree mode). I ask because there are a bunch of handsets that allow paging/broadcasting through their speakerphone mechanisms. This could possibly work in your scenario, even if all handsets don't support paging (it would generally be loud enough to hear, depending on the size of the office). Cheers, AR -- -- Alex Robar [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to notify an event to every user
2008/9/21 Gordon Henderson [EMAIL PROTECTED][EMAIL PROTECTED] On Sun, 21 Sep 2008, Olivier wrote: Hi, I've got this case : When the last staff member is about to leave and lock offices, he would like to notify everybody Offices are about to be closed so that (s)he wouldn't lock anybody in. Which is the smartest way to do it ? I thought of either : 1. sending an SMS, 2. calling every extension 1 by 1 with a pre-recorded message, ... SMS is fine but it is desired that members shouldn't find Were about to close message when they arrive in the morning. Calling everybody might take a long time and difficult to tame (as people forward calls here and there). Any idea ? Arrange the building to have a master lights off switch. Push it, then wait for the screams. This was used in a place I worked some years back. Use the intercom/page functions on the phones you have - dial the page button - say Anyone left? and if no-one screams, then turn the lights out. Arrange a ring-time for night bells - ring it. Wait for the screams. Many ways. The simplest might just be to use a lock system that can be opened from the inside. (alarms not withstanding) Nice ideas ! Thanks Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to notify an event to every user
2008/9/22 Steve Totaro [EMAIL PROTECTED] As said before, paging would work well. A walk through of the building would be helpful too. In this case, walking all the way through the building is not possible (way too long). Although rash, pull the fire alarm, making sure to remove the connections to the fire department. Killing the power shortly after would certainly help providing you have lighted emergency exit signs. You could leave a sign on the door when you lock it with a number to call to get out, this could be done via Asterisk with right kind of automagic door lock, otherwise, stick around and get ready to answer calls. Nice idea : I haven't thought about that one. There are many buildings with stairwells that open to the stairs but will not open from the stairs, maybe if you give more details, a good solution could be devised. A few cans of tear gas can clear an area post haste ;-) Curious, are you shutting down your company without giving notice? Not at the moment but, just in case, I will try to use tear gas, first ;-) I feel that if an employee should give two weeks notice, so should the employer. If not, you could simply install a key card system for at least the main door, or a motion sensor that activates a servo in the door locking mechanism. Either way, it should be fairly cheap. Thanks, Steve Totaro On Sun, Sep 21, 2008 at 5:30 PM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Sun, 21 Sep 2008, Olivier wrote: Hi, I've got this case : When the last staff member is about to leave and lock offices, he would like to notify everybody Offices are about to be closed so that (s)he wouldn't lock anybody in. Which is the smartest way to do it ? I thought of either : 1. sending an SMS, 2. calling every extension 1 by 1 with a pre-recorded message, ... SMS is fine but it is desired that members shouldn't find Were about to close message when they arrive in the morning. Calling everybody might take a long time and difficult to tame (as people forward calls here and there). Any idea ? Arrange the building to have a master lights off switch. Push it, then wait for the screams. This was used in a place I worked some years back. Use the intercom/page functions on the phones you have - dial the page button - say Anyone left? and if no-one screams, then turn the lights out. Arrange a ring-time for night bells - ring it. Wait for the screams. Many ways. The simplest might just be to use a lock system that can be opened from the inside. (alarms not withstanding) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to notify an event to every user
Not at the moment but, just in case, I will try to use tear gas, first ;-) I have found that with the right diet, teargas is not necessary. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to hangup a channel immediately so that it doesn't get charged on cell phone
In article [EMAIL PROTECTED],
Zeeshan Zakaria [EMAIL PROTECTED] wrote:
On my call back system, I have the script as follows:
[calback]
exten = s,1,NoOp(* STARTING CALLCHEAP\'S CALLBACK SYSTEM *)
exten = s,n,Set(CALL=${CALLERID(number)})
exten = s,n,Set(DESTINATION=myCallback.2000.1)
exten = s,n,Set(SLEEP=5)
exten = s,n,System(/var/lib/asterisk/bin/callback ${CALL} ${DESTINATION}
${SLEEP} )
exten = s,n,Hangup
The idea behind this system is that the script picks up the call, notes down
the caller's number, and hangs it immediately. Then the caller gets a call
back.
But what is happening is that cell phone callers are still being charged for
calling into this callback context.
How can I avoid this? I want cell phone users to not get charged for the
call back.
How does the incoming call get to calback,s,1 ? Is there another part of
the dialplan that receives the call and then jumps to here? If so, you
need to make sure that it doesn't call Answer(), nor any application that
might do an implicit answer.
Otherwise, please give more details about how the calls are delivered to
your system, and what you do with them right from the beginning.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN Simulator
You could buy toxic telco assets tied up in derivatives. mark morreny wrote: Hi, I have Asterisk setup to run on SS7, and I would like to test it out before getting the line from my telco. Is there any testing or simulation tool that I can buy to simulate a E1/SS7 link? Could anyone give some suggestions? Thanks alot for your help in advance. Regards, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [1.4.21.2] Checking that already off-hook?
Hello Here's the scenario in my extensions.conf: 1. Check that CID is available 2. If not, go off-hook, and prompt the caller to type their CID number 3. Whether it was sent directly by the telco or input by the caller, look up the CID number if the DB, and rewrite the CID name on the fly 4. In the main menu, if not already off-hook, go off-hook; Then, play a menu to choose an extension So if the user calls with a CID number unmasked, once in Step 4, I need to check if the FXO card is already off-hook before playing the menu. What's a reliable way to check for this? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transcoding G.729 files
Does anyone know of a utility I can use to transcode a group of files from G.729 format to something playable on a PC (GSM or WAV). I know I can convert them individually from the CLI, but I have quite a lot I need to do. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding G.729 files
SOX will do it if you install its G.729 format library. As far as converting a group of files, that's what scripting is for, i.e. for FILE in `find . -type f -name '*.g729'`; do NFILE=$(echo $FILE | sed 's/\.g729/\.wav/g') sox [some args] $FILE ... $NFILE ... done Thomas Kenyon wrote: Does anyone know of a utility I can use to transcode a group of files from G.729 format to something playable on a PC (GSM or WAV). I know I can convert them individually from the CLI, but I have quite a lot I need to do. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax with asterisk
Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] t38modem on OpenSuse
Hi All, is there anyone that tried to work with the t38modem project integrated with SIP through OPAL libraries in OpenSuse 10.2? I followed the cookbook at http://www.voip-info.org/wiki/index.php?page_id=5096 and I've a strange behavior. Firs of all when the t38modem starts, I've an error message that I think is related to some library not present in my current OpenSuse installation (but I'm not able to understand which library is still requiring, if anyone is able to help me to understand what's happening I'll be very happy to hear him). The message is: error loading avcodec - avcodec: cannot open shared object file: No such file or directory Running a ldd ./t38modem all seems ok. The next problem arises when I send faxes through an HT386 ATA and asterisk 1.4.20.1. Looking at the network traffic through ethereal I can see that the t38modem answer to the first INVITE message with a 100 TRYING message.. but it never send an ACK. At the same time, the t38modem is producing the log I've attached below (sorry for the long post). Any help is appreciated. Thank you. Marco Signorini 2008/09/22 23:53:39.395 Opal Liste...er:80b95c8 SIP PDU Received on udp$192.168.0.5:5060if=udp$192.168.0.5:6060 INVITE sip:[EMAIL PROTECTED]:6060 SIP/2.0 Date: Mon, 22 Sep 2008 21:53:39 GMT CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK2409d6cb User-Agent: Cadore 9 PBX From: Soggiorno2 sip:[EMAIL PROTECTED];tag=as6bf57b61 Call-ID: [EMAIL PROTECTED] Supported: replaces To: sip:[EMAIL PROTECTED]:6060 Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 508 Max-Forwards: 70 v=0 o=root 3222 3222 IN IP4 192.168.0.5 s=session c=IN IP4 192.168.0.5 t=0 0 m=audio 5018 RTP/AVP 0 97 3 8 112 5 10 7 18 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2008/09/22 23:53:39.400 Opal Liste...er:80b95c8 SDP Media session port=5018 2008/09/22 23:53:39.401 Opal Liste...er:80b95c8 SDP Adding media session with 11 formats 2008/09/22 23:53:39.402 Opal Liste...er:80b95c8 SDP Unknown media attribute silenceSupp:off - - - - 2008/09/22 23:53:39.405 Opal Liste...er:80b95c8 SIP Sending PDU on udp$192.168.0.5:5060if=udp$192.168.0.5:6060 SIP/2.0 100 Trying CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK2409d6cb From: Soggiorno2 sip:[EMAIL PROTECTED];tag=as6bf57b61 Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED]:6060 Contact: sip:[EMAIL PROTECTED]:6060;transport=udp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE,INFO,PING,PUBLISH Content-Length: 0 2008/09/22 23:53:39.408 Opal Liste...er:80b95c8 CallCreated Call[4] 2008/09/22 23:53:39.409 Opal Liste...er:80b95c8 MySIPEndPoint::CreateConnection for Call[4] 2008/09/22 23:53:39.409 Opal Liste...er:80b95c8 OpalCon Created connection Call[4]-EPsip[EMAIL PROTECTED] 2008/09/22 23:53:39.410 Opal Liste...er:80b95c8 RFC2833 Handler created 2008/09/22 23:53:39.411 Opal Liste...er:80b95c8 RFC2833 Handler created 2008/09/22 23:53:39.415 Opal Liste...er:80b95c8 OpalUDP Binding to interface: 192.168.0.5:5651 2008/09/22 23:53:39.416 Opal Liste...er:80b95c8 SIP Created transport udp$0.0.0.0if=udp$192.168.0.5:5651 2008/09/22 23:53:39.417 Opal Liste...er:80b95c8 OpalUDP Started connect to 192.168.0.5:6060 2008/09/22 23:53:39.418 Opal Liste...er:80b95c8 OpalUDP Connect on pre-bound interface: 192.168.0.5 2008/09/22 23:53:39.419 Opal Liste...er:80b95c8 PWLib Created thread 0x80e1690 SIP Transport:%x 2008/09/22 23:53:39.420 Opal Liste...er:80b95c8 SIP Created connection. 2008/09/22 23:53:39.421 Opal Liste...er:80b95c8 SIP Queueing PDU: 102 INVITE sip:[EMAIL PROTECTED]:6060 2008/09/22 23:53:39.422 Opal Liste...er:80b95c8 PWLib Created thread 0x80e3190 SIP Handler:%x 2008/09/22 23:53:39.422 Opal Liste...er:80b95c8 OpalTransport clean up on termination 2008/09/22 23:53:39.423 Opal Liste...er:80b95c8 OpalUDP Close 2008/09/22 23:53:39.423 Opal Liste...er:80b95c8 OpalDeleted transport udp$192.168.0.5:5060if=udp$192.168.0.5:6060 2008/09/22 23:53:39.556 Opal Liste...er:80b95c8 Listen Waiting on UDP packet on udp$192.168.0.5:6060 2008/09/22 23:53:39.557 SIP Transp...rt:80e1690 PWLib Started thread 0x80e1690 SIP Transport:80e1690 2008/09/22 23:53:39.557 SIP Transp...rt:80e1690 SIP Read thread started. 2008/09/22 23:53:39.558 SIP Transp...rt:80e1690 SIP Waiting for PDU on udp$192.168.0.5:6060if=udp$192.168.0.5:5651 2008/09/22 23:53:39.559 SIP Handle...er:80e3190 PWLib Started thread 0x80e3190 SIP Handler:80e3190 2008/09/22 23:53:39.559 SIP Handle...er:80e3190 SIP PDU handler thread started. 2008/09/22 23:53:39.560 SIP
[asterisk-users] Registration by IP address
Dear All, I'm using a2billing interface with asterisk in order to bill all calls flowing through my PBX... I need to prevent my customers to use the same extension from different IP addresses so I created a new extension under extensions.conf as follow: [michofr] type=peer username=michofr accountcode=4197464352 regexten=michofr callerid=11 amaflags=billing secret=123456 nat=yes dtmfmode=RFC2833 qualify=yes canreinvite=yes disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 host=192.168.0.164 context=a2billing regseconds=0 cancallforward=yes When trying to registr I'm getting 403 Forbidden...I think it's a domain issue under sip.conf fileCan someone help me in that please? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI and prepaid billing
Hi All; Did anyone do an prepaid billing application via AGI? I would like to know if that is possible. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_misdn troubles
Hello I have just set up Asterisk Asterisk 1.4.21.2 on a CentOS 5.2 machine. I am using the OpenVox B200P ISDN card. My problem is that even though chan_misdn module seems to be loaded correctly with Asterisk (I can see it using 'module show' command) the misdn commands are not available to me in the CLI so I cannot tell if my box is correctly interfacing with the ISDN card Any ideas what can be going wrong ? My installation procedure looked like this : cd /usr/src/ wget http://www.misdn.org/downloads/mISDN.tar.gz wget http://www.misdn.org/downloads/mISDNuser.tar.gz tar xzf mISDN.tar.gz tar xzf mISDNuser.tar.gz cd ../mISDN-1_1_7_2/ make install cd ../mISDNuser-1_1_7_2/ make install cd asterisk-1.4.21.2/ make menuconfig (I chose chan_misdn) make; make install; ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_misdn troubles
Tuesday, September 23, 2008, 11:57:00 AM, Thanos wrote: Hello I have just set up Asterisk Asterisk 1.4.21.2 on a CentOS 5.2 machine. I am using the OpenVox B200P ISDN card. My problem is that even though chan_misdn module seems to be loaded correctly with Asterisk (I can see it using 'module show' command) the misdn commands are not available to me in the CLI so I cannot tell if my box is correctly interfacing with the ISDN card Any ideas what can be going wrong ? ... cd ../mISDN-1_1_7_2/ What kernel version you use? Newer linux kernels (2.6.24) works only with new (and beta) 1.1.8 misdn. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_misdn troubles
Hi! I'm also still new to this. but perhaps: Did you do make examples, or did you have an earlier asterisk installation, so the configuration files were present? If so did you make sure, that you mISDN card was properly configured, using: 1. misdn-init scan 2. misdn-init config 3. misdn-init start Or probably reboot your machine to make sure the mISDN-kernel-site (misdn.org) are properly start and initialised. Make sure you have: 1. /dev/mISDN 2. all the modules loaded: lsmod Then edit the misdn.conf in /etc/asterisk and make sure it is setup to your needs (ports, msns, jitter, etc. I remember, that I had a lot of pain to get it working on my machine. Mostly due to kernel problems (linux 2.6.2x). I hope this helps. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4.21.2] Checking that already off-hook?
Hi! I wouldn't know a proper way to check for off-hook. But, couldn't you change your dialplan? 1. answer the call 2. check for CID 3. branch with a gotoif 4. Enter CID 5. Look up CID in your DB and whatever 6. Playback the mainmenu welcome [go on] Something like this? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4.21.2] Checking that already off-hook?
On Tue, 23 Sep 2008 12:23:28 +0200 (CEST), Julien Claassen [EMAIL PROTECTED] wrote: I wouldn't know a proper way to check for off-hook. But, couldn't you change your dialplan? Thanks for the suggestion, and this is how the script works now, but since most customers do call with CID enabled, I'd like to send a broadcast on the LAN to display this information on everyone's PC before Asterisk goes off-hook and does its spiel. Isn't there a way to check the status an FXO card is in? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and prepaid billing
Hi Bilal, Yes it is definitely possible. And I've done it myself for a couple of our clients. Does that answer your two questions? cheers - Ben. --- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote: From: bilal ghayyad [EMAIL PROTECTED] Subject: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com Date: Tuesday, September 23, 2008, 9:52 AM Hi All; Did anyone do an prepaid billing application via AGI? I would like to know if that is possible. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
Hi Erik, we use an ATA device connected to the fax machine. If you want to receive faxes, since Asterisk fax detection is not reliable, use one DID to link it directly to the ATA: you lose a number but you gain a fully-working fax! Giorgio Incantalupo. Erik Haider Forsen wrote: Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_misdn troubles
On Tue, Sep 23, 2008 at 1:19 PM, Julien Claassen [EMAIL PROTECTED] wrote: Hi! I'm also still new to this. but perhaps: Did you do make examples, or did you have an earlier asterisk installation, so the configuration files were present? If so did you make sure, that you mISDN card was properly configured, using: 1. misdn-init scan 2. misdn-init config 3. misdn-init start Or probably reboot your machine to make sure the mISDN-kernel-site (misdn.org) are properly start and initialised. Make sure you have: 1. /dev/mISDN 2. all the modules loaded: lsmod Then edit the misdn.conf in /etc/asterisk and make sure it is setup to your needs (ports, msns, jitter, etc. I remember, that I had a lot of pain to get it working on my machine. Mostly due to kernel problems (linux 2.6.2x). I hope this helps. Kindest regards Julien I have misdn running from chkconfig for my run level /dev/mISDN exists and I do see the ISDN modules loaded : mISDN_core 78720 6 ISDN_dsp,hfcmulti,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1 Can you send me a sample of your misdn.conf file maybe that is the actual problem because I am pretty much at a loss at the moment As for the kernel I use : 2.6.18-92.1.10.el5.centos.plus thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_misdn troubles
On Tue, Sep 23, 2008 at 1:19 PM, Gergo Csibra [EMAIL PROTECTED] wrote: Tuesday, September 23, 2008, 11:57:00 AM, Thanos wrote: Hello I have just set up Asterisk Asterisk 1.4.21.2 on a CentOS 5.2 machine. I am using the OpenVox B200P ISDN card. My problem is that even though chan_misdn module seems to be loaded correctly with Asterisk (I can see it using 'module show' command) the misdn commands are not available to me in the CLI so I cannot tell if my box is correctly interfacing with the ISDN card Any ideas what can be going wrong ? ... cd ../mISDN-1_1_7_2/ What kernel version you use? Newer linux kernels (2.6.24) works only with new (and beta) 1.1.8 misdn. -- Best regards, Gergomailto:[EMAIL PROTECTED] Using 2.6.18-92.1.10.el5.centos.plus kernel so I suppose that should be OKs. The modules are correctly loading ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
Fair enough, I did not attend bootcamp, and I passed the dcap at Astricon 2004. My opinion was based on a number of questions in the written exam that I felt had nothing to do with either Asterisk or integration of Asterisk into a customer site. My assumption therefore was that those questions covered content taught in the Bootcamp. I am happy to stand corrected on the matter. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Brentano Sent: Monday, 22 September 2008 1:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium training course I would also disagree that the written exam is biased towards people who attended the training. I attended a Bootcamp earlier this year and thought I was fully prepared to pass the dCAP. Especially since I already had real-world Asterisk experience. But the written exam covered material that we hadn't even discussed in class, some stuff that was in the book, and other that I was totally lost on. I passed the practical with a near perfect score, but fell just short of passing the written. IMHO, the written portion needs to be re-evaluated. What I think needs to change is de-coupling the dCAP exam from the Bootcamp class. I'll likely never retake the dCAP exam since Digium doesn't offer the Bootcamp in my area (Portland) and I can't go to a local testing facility (New Horizons, et al.) and do the exam. It would cost me well beyond the $300 to take the exam after factoring in travel costs and time spent away from work. Also, the problem with the dCAP being coupled to the Bootcamp is that it gives you the false impression that the Bootcamp prepares you to pass the dCAP and that is completely *not true*. In my Bootcamp class of 9 only 4 took the dCAP. Our own instructor said it took him 3 tries to pass! If this isn't going to change, then the dCAP should be changed so that the Bootcamp *does* prepare you to pass. And similarly, Digium should then also offer less expensive (at least, less than $3K) self-study materials or online training that also offers similar training without having to be present at the Bootcamp That way someone could elect to train at their own schedule and later coordinate to drop-in on the last day of a Bootcamp session and take the dCAP. - Chris On Sep 21, 2008, at 9:15 AM, Tilghman Lesher wrote: On Thursday 18 September 2008 20:56:58 Craig Guy wrote: I felt at the time the written portion was heavily biased towards people who had done the training - in fact I would go so far as to say that it was designed specifically to discriminate against people who had not attended the official training. I'd have to disagree with that, having taken the written portion without having attended the bootcamp, and I got one of the highest scores of the people there that day. Included was one question that I believe I was the only that day to have gotten right. Of course, I had the written the application upon which that question was based, so I had an unfair advantage, I suppose. Other than that question, though, I'd have to say that the written portion highly favored the person with a well-rounded set of experiences with Asterisk. However, the test has been revised since I have taken it, and Jared assures me that some of the more tricky questions have been removed, so the written portion may be easier nowadays. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding G.729 files
Alex Balashov wrote: SOX will do it if you install its G.729 format library. As far as converting a group of files, that's what scripting is for, i.e. for FILE in `find . -type f -name '*.g729'`; do NFILE=$(echo $FILE | sed 's/\.g729/\.wav/g') sox [some args] $FILE ... $NFILE ... done Thanks, didn't know sox could support g.729. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to notify an event to every user
2008/9/23 Paul Hales [EMAIL PROTECTED] Not at the moment but, just in case, I will try to use tear gas, first ;-) I have found that with the right diet, teargas is not necessary. That interesting to know. Maybe we should open a new thread on that and let everyone contribute ;-) PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
2008/9/23 Giorgio Incantalupo [EMAIL PROTECTED] Hi Erik, we use an ATA device connected to the fax machine. If you want to receive faxes, since Asterisk fax detection is not reliable Hi, Which fax detection did you used, then ? , use one DID to link it directly to the ATA: you lose a number but you gain a fully-working fax! Giorgio Incantalupo. Erik Haider Forsen wrote: Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
I have an TDM800P+ata+fax and work fine. This setup take 5 min. The best solution must be hylax fax + asterisk. But you need an asterisk specialist to make the setup and take more time. With this solution you can send fax and receive fax in your inbox and reduce toner/papper costs. Regards, Luis Morales On Wed, Sep 24, 2008 at 4:21 AM, Erik Haider Forsen [EMAIL PROTECTED] wrote: Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension registration
Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
Hi Olivier, We DO NOT use faxdetect because it does not work properly. That's why we link a PRI DID to it, so when people call that DID the fax machine gets direct fax data without passing thru faxdetection. Giorgio Incantalupo. Olivier wrote: 2008/9/23 Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi Erik, we use an ATA device connected to the fax machine. If you want to receive faxes, since Asterisk fax detection is not reliable Hi, Which fax detection did you used, then ? , use one DID to link it directly to the ATA: you lose a number but you gain a fully-working fax! Giorgio Incantalupo. Erik Haider Forsen wrote: Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to notify an event to every user
Olivier a écrit : Hi, Good day I've got this case : When the last staff member is about to leave and lock offices, he would like to notify everybody Offices are about to be closed so that (s)he wouldn't lock anybody in. Which is the smartest way to do it ? I thought of either : 1. sending an SMS, 2. calling every extension 1 by 1 with a pre-recorded message, ... SMS is fine but it is desired that members shouldn't find Were about to close message when they arrive in the morning. Calling everybody might take a long time and difficult to tame (as people forward calls here and there). We setup something like this with Snom and auto answer. You can call all phones in a time. You even can use meetme stuff to complete the setup (only listening). -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension registration
Make host=dynamic. Atenciosamente, Vinícius Fontes Núcleo de Tecnologias Convergentes Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Convergent Technologies Core Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - michel freiha [EMAIL PROTECTED] escreveu: Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host= 192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP= 192.168.0.164 , I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension registration
michel freiha a écrit : Hi all, Hi I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Remove the secret or put host=dynamic. You can't register when you define the host IP address -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension registration
michel freiha wrote: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 http://192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 You've forgotten nat=yes. You'll also want to specify a context on your mailbox line. (i.e. [EMAIL PROTECTED]) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension registration
With host=dynamic it's working fine...I need to force the user to use his extension from one IP address and not from different IP addresses Regards On Tue, Sep 23, 2008 at 3:40 PM, Vinícius Fontes [EMAIL PROTECTED]wrote: Make host=dynamic. Atenciosamente, Vinícius Fontes Núcleo de Tecnologias Convergentes Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Convergent Technologies Core Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - michel freiha [EMAIL PROTECTED] escreveu: Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host= 192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP= 192.168.0.164 , I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension registration
On Sep 23, 2008, at 8:40 AM, Vinícius Fontes wrote: Make host=dynamic. Also, set nat=yes Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host= 192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 Fred Posner [EMAIL PROTECTED] Using VoIP? SIP:[EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension registration
If I make host=dynamic, then the customer will be able to register on my asterisk server from any IP address...What I need is to force the User to register on asterisk from a specific IP address like 192.168.0.164...How this could be done? Regards On Tue, Sep 23, 2008 at 3:52 PM, Fred Posner [EMAIL PROTECTED] wrote: On Sep 23, 2008, at 8:40 AM, Vinícius Fontes wrote: Make host=dynamic. Also, set nat=yes Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host= 192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 Fred Posner [EMAIL PROTECTED] Using VoIP? SIP: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension registration
The user won't need to register at all, registration is only good if the ip address changes. Much simpler that way. Just put host=the ip address you want on Tuesday 09/23/2008 michel freiha([EMAIL PROTECTED]) wrote If I make host=dynamic, then the customer will be able to register on my asterisk server from any IP address...What I need is to force the User to register on asterisk from a specific IP address like 192.168.0.164...How this could be done? Regards On Tue, Sep 23, 2008 at 3:52 PM, Fred Posner [EMAIL PROTECTED] wrote: On Sep 23, 2008, at 8:40 AM, Vinícius Fontes wrote: Make host=dynamic. Also, set nat=yes Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host= 192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 Fred Posner [EMAIL PROTECTED] Using VoIP? SIP: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users div dir=ltrdivIf I make host=dynamic, then the customer will be able to register on my asterisk server from any IP address...What I need is to force the User to register on asterisk from a specific IP address like 192.168.0.164...How this could be done?/div divnbsp;/div divRegardsbrbr/div div class=gmail_quoteOn Tue, Sep 23, 2008 at 3:52 PM, Fred Posner span dir=ltrlt;a href=mailto:[EMAIL PROTECTED][EMAIL PROTECTED]/agt;/span wrote:br blockquote class=gmail_quote style=PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid div style=WORD-WRAP: break-word divspan style=FONT-WEIGHT: normal; FONT-SIZE: 12px; WORD-SPACING: 0px; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; LINE-HEIGHT: normal; FONT-STYLE: normal; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate; FONT-VARIANT: normal div style=WORD-WRAP: break-wordspan style=FONT-WEIGHT: normal; FONT-SIZE: 12px; WORD-SPACING: 0px; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; LINE-HEIGHT: normal; FONT-STYLE: normal; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate; FONT-VARIANT: normal div style=WORD-WRAP: break-wordspan style=FONT-WEIGHT: normal; FONT-SIZE: 12px; WORD-SPACING: 0px; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; LINE-HEIGHT: normal; FONT-STYLE: normal; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate; FONT-VARIANT: normal div style=WORD-WRAP: break-wordspan style=FONT-WEIGHT: normal; FONT-SIZE: 12px; WORD-SPACING: 0px; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; LINE-HEIGHT: normal; FONT-STYLE: normal; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate; FONT-VARIANT: normal div style=WORD-WRAP: break-wordspan style=FONT-WEIGHT: normal; FONT-SIZE: 12px; WORD-SPACING: 0px; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; LINE-HEIGHT: normal; FONT-STYLE: normal; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate; FONT-VARIANT: normal div style=WORD-WRAP: break-wordspan style=FONT-WEIGHT: normal; FONT-SIZE: 12px; WORD-SPACING: 0px; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; LINE-HEIGHT: normal; FONT-STYLE: normal; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate; FONT-VARIANT: normalspan style=FONT-WEIGHT: normal; FONT-SIZE: 12px; WORD-SPACING: 0px; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; LINE-HEIGHT: normal; FONT-STYLE: normal; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate; FONT-VARIANT: normal div style=WORD-WRAP: break-word divbr/div/div/span/span/div/span/div/span/div/span/div/span/div/span/div div divOn Sep 23, 2008, at 8:40 AM, Vinícius Fontes wrote:/divbr blockquote type=cite divMake host=dynamic.brbrbr/div/blockquote divbr/div divAlso, set nat=yes/div div class=Ih2E3dbr blockquote type=cite div blockquote type=citeHi all,br/blockquote blockquote type=citenbsp;br/blockquote blockquote type=citeI have the below extension defined under sip.conf:br/blockquote blockquote type=citenbsp;br/blockquote blockquote type=cite[2203]br/blockquote blockquote type=citetype=friendbr/blockquote blockquote type=citeusername=2203br/blockquote blockquote type=citesecret=123456br/blockquote blockquote type=citehost= a href=http://192.168.0.164/; target=_blank192.168.0.164/abr/blockquote blockquote type=citemailbox=2203br/blockquote blockquote
Re: [asterisk-users] Extension registration
Is there a way to register to asterisk only from a specific IP address, which mean the customer can use his extension only from one IP address? Regards On Tue, Sep 23, 2008 at 3:49 PM, Administrator TOOTAI [EMAIL PROTECTED]wrote: michel freiha a écrit : Hi all, Hi I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Remove the secret or put host=dynamic. You can't register when you define the host IP address -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
Hey Erik, You can also check out pika technologies which supply chan_pika. This comes with a fax application that will let you do your faxes in asterisk (even using non-pika boards). Works pretty good... pikatechnologies.com mattm On Tue, Sep 23, 2008 at 4:51 AM, Erik Haider Forsen [EMAIL PROTECTED] wrote: Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
Hi Giorgio, Thanks for your answer. Your setup is exactly what we're thinking of. We have 1100 DID's, so that shouldn't be a problem at all. Which ATA box are you using? Erik On Sep 23, 2008, at 2:06 PM, Giorgio Incantalupo wrote: Hi Olivier, We DO NOT use faxdetect because it does not work properly. That's why we link a PRI DID to it, so when people call that DID the fax machine gets direct fax data without passing thru faxdetection. Giorgio Incantalupo. Olivier wrote: 2008/9/23 Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi Erik, we use an ATA device connected to the fax machine. If you want to receive faxes, since Asterisk fax detection is not reliable Hi, Which fax detection did you used, then ? , use one DID to link it directly to the ATA: you lose a number but you gain a fully-working fax! Giorgio Incantalupo. Erik Haider Forsen wrote: Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
Hi Matthew, Thanks for your suggestion. The problem is that most of our users would not feel comfortable with using software fax solutions. So we will have to stick with the old fax machine. Our reception takes care of the fax machine, receiving and sending faxes. This one fax is shared by ~ 600 employees. Erik On Sep 23, 2008, at 3:25 PM, Matthew Marion wrote: Hey Erik, You can also check out pika technologies which supply chan_pika. This comes with a fax application that will let you do your faxes in asterisk (even using non-pika boards). Works pretty good... pikatechnologies.com mattm On Tue, Sep 23, 2008 at 4:51 AM, Erik Haider Forsen [EMAIL PROTECTED] wrote: Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6b9 Audio Issue
To close the loop on this I have found that this appears to no longer be an issue since I moved to 1.6rc6. Mark Michelson wrote: MFH wrote: I'm noticing in 1.6 Beta 9 that on outgoing calls I get a brief audio drop when the audio starts on the other end of the call. So basically I hear the first word, miss the second word and then hear the rest fine. I've noticed this after calling multiple locations and getting some recording on the other end. The origin of the outbound channel is always SIP but the asterisk to PSTN could be SIP or IAX. Anyone else? MARK. One difference between Asterisk 1.6.0 and previous versions is that when a channel answers, there is a built-in 500 ms delay so that media has time to be set up. This may be what you are experiencing. There was a bug reported recently that was traced back to this delay. In the next 1.6.0 tarball, the delay will behave slightly differently, although I doubt it will be noticeable for the situation you have described. The bug I refer to is: http://bugs.digium.com/view.php?id=12924 Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension registration
On Sep 23, 2008, at 9:03 AM, michel freiha wrote: Is there a way to register to asterisk only from a specific IP address, which mean the customer can use his extension only from one IP address? host=192.168.0.164 Yes, use the external IP that the client is sending you instead of the NAT address. Also, add nat=yes if you're doing that. Fred Posner [EMAIL PROTECTED] Tel: +1 (212) 937-7844 x501 www.teamforrest.com Using VoIP? SIP:[EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension registration
The NAT Network Address Translation is a layer three protocol... it encapsulates the End user's IP address with the router's IP address... so your Asterisk is not recognizing the IP address of the end user.. if you are insisting on using the HOST option with a specific IP.. maybe you should use the Global Nat Address of the router your end user is located behind it.. and if you don't want your client to use any other PC on the same nat.. then i think you need to permit the Local IP address of the user with the : host=your end user's router's real ip deny=0.0.0.0/0.0.0.0 permit=192.168.x.y/255.255.255.255 that will force the Global ip to register with the local ip.. on my asterisk i have several users registering with different Agents.. from one nat one global ip address but several private ips.. my asterisk recognizes the private ip AFTER the real IP does the registration.. i'm not sure of that though.. but it's worth trying! let me know if this answer was saticfying to you . regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Tue, 23 Sep 2008 16:00:41 +0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Extension registration If I make host=dynamic, then the customer will be able to register on my asterisk server from any IP address...What I need is to force the User to register on asterisk from a specific IP address like 192.168.0.164...How this could be done? Regards On Tue, Sep 23, 2008 at 3:52 PM, Fred Posner [EMAIL PROTECTED] wrote: On Sep 23, 2008, at 8:40 AM, Vinícius Fontes wrote: Make host=dynamic. Also, set nat=yes Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host= 192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 Fred Posner [EMAIL PROTECTED] Using VoIP? SIP: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get more out of the Web. Learn 10 hidden secrets of Windows Live. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
ATAs work OK I guess, just make sure to use a loss less codec such as ULAW. Personally, I would install a single port FXS card in the Asterisk server to avoid any IP transport. You may have to mess with the gains to get it working very well, but nothing beats your existing POTS line. If you had many physical fax machines and a spare T1 port, I would suggest a channel bank, this has worked on par with a POTS line in my experience, providing you are getting PSTN access through a T1/E1. Thanks, Steve Totaro On Tue, Sep 23, 2008 at 9:32 AM, Erik Haider Forsen [EMAIL PROTECTED] wrote: Hi Matthew, Thanks for your suggestion. The problem is that most of our users would not feel comfortable with using software fax solutions. So we will have to stick with the old fax machine. Our reception takes care of the fax machine, receiving and sending faxes. This one fax is shared by ~ 600 employees. Erik On Sep 23, 2008, at 3:25 PM, Matthew Marion wrote: Hey Erik, You can also check out pika technologies which supply chan_pika. This comes with a fax application that will let you do your faxes in asterisk (even using non-pika boards). Works pretty good... pikatechnologies.com mattm On Tue, Sep 23, 2008 at 4:51 AM, Erik Haider Forsen [EMAIL PROTECTED] wrote: Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension registration
Fred, The context should stay friend or i should change it to another thing? Regards On Tue, Sep 23, 2008 at 4:59 PM, Fred Posner [EMAIL PROTECTED] wrote: On Sep 23, 2008, at 9:03 AM, michel freiha wrote: Is there a way to register to asterisk only from a specific IP address, which mean the customer can use his extension only from one IP address? host=192.168.0.164 Yes, use the external IP that the client is sending you instead of the NAT address. Also, add nat=yes if you're doing that. Fred Posner [EMAIL PROTECTED] Tel: +1 (212) 937-7844 x501 www.teamforrest.com Using VoIP? SIP: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
Just to clarify, pika's chan_pika is their asterisk channel driver for their
hardware. The two fax applications, app_pikarxfax.so and app_pikatxfax.so
are fax modules for asterisk to implement fax in your dialplan:
exten = 5001,1,Answer(1000);
exten = 5001,n,Set(LOCALSTATIONID=123456789)
exten = 5001,n,Set(LOCALHEADERINFO=PIKARxFax Test Page %P Time: %H:%M To:
%l From: %r)
exten = 5001,n,Set(FAXFILE=/tmp/pikafax-${UNIQUEID}.tif)
exten = 5001,n,PIKARxFax(${FAXFILE})
you'd still need hardware to actually get the faxes to asterisk...
On Tue, Sep 23, 2008 at 9:32 AM, Erik Haider Forsen [EMAIL PROTECTED] wrote:
Hi Matthew,
Thanks for your suggestion. The problem is that most of our users
would not feel comfortable with using software fax solutions. So we
will have to stick with the old fax machine.
Our reception takes care of the fax machine, receiving and sending
faxes. This one fax is shared by ~ 600 employees.
Erik
On Sep 23, 2008, at 3:25 PM, Matthew Marion wrote:
Hey Erik,
You can also check out pika technologies which supply chan_pika.
This comes with a fax application that will let you do your faxes in
asterisk (even using non-pika boards). Works pretty good...
pikatechnologies.com
mattm
On Tue, Sep 23, 2008 at 4:51 AM, Erik Haider Forsen
[EMAIL PROTECTED] wrote:
Hi!
I'm new to this list. I tried to search the list archive for a
solution on my current setup, but couldn't find any.
We have an asterisk connected directly to the PSTN with 2 E1 lines
through a Sangoma A102d interface. We also have a regular FAX machine.
My question is how to get the fax service handled by asterisk? I want
to cancel the analog line I have for the FAX machine today.
What would be the best solution? Fax machine and asterisk is on the
same LAN, not much load, with high end switches etc. Can I expect good
results with using our existing FAX machine, connected to asterisk
through an ATA box?
Best Regards,
Erik
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
Forgot to mention (I think) that though the chan_pika driver is for pika
hardware, the two fax apps for asterisk work with 3rd party hardware as
well, so you don't actually need the pika cards (that's why it's nice and
easy)
On Tue, Sep 23, 2008 at 10:23 AM, Matthew Marion [EMAIL PROTECTED]wrote:
Just to clarify, pika's chan_pika is their asterisk channel driver for
their hardware. The two fax applications, app_pikarxfax.so and
app_pikatxfax.so are fax modules for asterisk to implement fax in your
dialplan:
exten = 5001,1,Answer(1000);
exten = 5001,n,Set(LOCALSTATIONID=123456789)
exten = 5001,n,Set(LOCALHEADERINFO=PIKARxFax Test Page %P Time: %H:%M To:
%l From: %r)
exten = 5001,n,Set(FAXFILE=/tmp/pikafax-${UNIQUEID}.tif)
exten = 5001,n,PIKARxFax(${FAXFILE})
you'd still need hardware to actually get the faxes to asterisk...
On Tue, Sep 23, 2008 at 9:32 AM, Erik Haider Forsen [EMAIL PROTECTED]wrote:
Hi Matthew,
Thanks for your suggestion. The problem is that most of our users
would not feel comfortable with using software fax solutions. So we
will have to stick with the old fax machine.
Our reception takes care of the fax machine, receiving and sending
faxes. This one fax is shared by ~ 600 employees.
Erik
On Sep 23, 2008, at 3:25 PM, Matthew Marion wrote:
Hey Erik,
You can also check out pika technologies which supply chan_pika.
This comes with a fax application that will let you do your faxes in
asterisk (even using non-pika boards). Works pretty good...
pikatechnologies.com
mattm
On Tue, Sep 23, 2008 at 4:51 AM, Erik Haider Forsen
[EMAIL PROTECTED] wrote:
Hi!
I'm new to this list. I tried to search the list archive for a
solution on my current setup, but couldn't find any.
We have an asterisk connected directly to the PSTN with 2 E1 lines
through a Sangoma A102d interface. We also have a regular FAX machine.
My question is how to get the fax service handled by asterisk? I want
to cancel the analog line I have for the FAX machine today.
What would be the best solution? Fax machine and asterisk is on the
same LAN, not much load, with high end switches etc. Can I expect good
results with using our existing FAX machine, connected to asterisk
through an ATA box?
Best Regards,
Erik
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI incoming call forward / call redirect
Good morning,
I have a Bell Canada PRI here (switchtype=national) and I am trying to perform
a call-forward-unconditional on one of the DIDs.
The idea is that when DID 5551234 receives a call, Asterisk redirects it back
out the same PRI to some external number.
This is simple enough to do with something along these lines:
[PRI]
exten = 5551234,1,Set(CALLERID(RDNIS)=${EXTEN})
exten = 5551234,n,Dial(Zap/g1/5556789)
This is a brute-force approach but there are two problems:
1) it's not a true call forward
2) RDNIS does not appear to be getting set (i.e. the remote box with 5556789
as a DID does not seem to see RDNIS
I'm not overly concerned about 2BCT capability at this time (it *is* talking
to a 5ESS although I'm not sure if Asterisk will attempt 2BCT with national-2
switchtype) but it is important to be able to retrieve RDNIS, as the hope is
to redirect a number of DIDs to one external number, and have the external
number see which the original number was through RDNIS.
I had this working great the other way -- some external POTS number
call-forwarded with *72 to a DID on this PRI, the DID saw RDNIS just fine,
but now I'm trying to go the other way round.
Any ideas? This is Asterisk 1.4.18.
Regards,
Andrew
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension registration
Fred, The context should stay friend or i should change it to another thing? Regards This would depend on what you want that user to be able to do... Here's a good source to learn the differences: http://www.voip-info.org/wiki/view/Asterisk+sip+type Fred Posner [EMAIL PROTECTED] Using VoIP? SIP:[EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: more on Free World Dialup groups and FWDLive
FYI It looks like FWD is looking for value added service ideas for free as a volunteer. I think it will fail but we shall see. I really don't get the nerve of them (Free World Dialup has changed it's name to FWD) to ask for free ideas and development on a non-free service. Maybe if they can come up with a killer app and people will adopt it, then it might work, but then again, people still cling to their analog FAX machines Thanks, Steve Totaro -- Forwarded message -- From: Daniel Berninger [EMAIL PROTECTED] Date: Tue, Sep 23, 2008 at 10:39 AM Subject: more on Free World Dialup groups and FWDLive To: [EMAIL PROTECTED] Hello, We are looking for group leaders and topic ideas for the FWD voice analog of Yahoo!Groups - FWDLive. The exact approach to FWDLive remains a work in progress. We know FWDLive should offer SIP enabled group conversations along the lines of an open protocol version of Talkshoe. We may end up limiting the size and access to groups to avoid the sort of disruptive participants that led to the demise of Skypecasts. A prototype of process for creating groups will get posted to FWDWiki: 1) pick a topic and write short summary 2) pick a time to run the call, post to the schedule, request conference code 3) dial into the group at the appointed time Jeff Pulver will host a call today at 2:00 ET to discuss FWDLive topic ideas. Reply to this note if your are interested in joining the call with Jeff or volunteering as a group leader. I also attached a VoIP Planet article below that provides more details on why FWD moved to paid membership. Best regards, Dan ... Daniel Berninger CEO, FWD fwd: 12908 v: +1.202.250.3838 e: [EMAIL PROTECTED] w: www.freeworlddialup.com http://www.voipplanet.com/news/article.php/3767266 Free World Dialup No Longer Free August 22, 2008 By Jeff Goldman FWD, formerly known as Free World Dialup, will next month start charging a mandatory subscription fee of $30 per year, as part of a larger plan to reinvent itself as what the company calls a 'Communication ISP.' This follows FWD's introduction a year ago of an optional $30-a-year membership plan. According to FWD CEO Daniel Berninger, the mandatory fee was simply a logical next step. The voluntary one gave us the confidence to do the required one... it was pretty successful, so what we ended up figuring out over the year was that we wanted to be able to fund ourselves enough so that we wouldn't have to do any kind of PSTN funding, like selling DIDs, he says. And that, Berninger says, is really the point. After a decade, VoIP hasn't reached its potential—it basically is an on-ramp to the telephone network, and doesn't do anything else, he says. People have experimented with things, but for the most part, all the revenue models of [companies like] Skype and JAJAH... have something to do with extracting money based on usage charges and giving people access to the telephone network. Instead, Berninger wants to turn FWD into a Communication ISP, an idea he introduced in a blog post earlier this month in which he argued that Interconnection with the telephone network shuts out the possibility of creativity... Content is limited to those uses justified in the context of the per minute cost of telephone service. And so the Communication ISP is intended to be a pure SIP offering, free of the PSTN and its inherent restrictions. For your regular ISP, you pay them a monthly fee and they attach your computer to the Internet... we want to be the same thing, in that you buy a communication device, a SIP VoIP device, and you go to a Communication ISP and get the thing on the Internet... and from there, you build applications and create new value, he says. So we're thinking about this like an entire ecosystem. To compete with the dominance of the PSTN, Berninger says, VoIP needs to differentiate itself better, not only with things like video and wideband audio, but also with a whole new range of as-yet-unknown applications. The hard part of the argument is this bootstrap problem—in other words, how do we get from where we are, not knowing what the applications are and not having anybody with capable devices, to scale? he says. The parallel, of course, would be the early days of the Internet. When it started, there was a very small audience and very limited content, but it did have global termination for the same price... and it created the virtuous cycle of content attracting more audience and audience attracting more content—and the next thing you know, the thing's growing tenfold a year, he says. To begin with, Berninger says, the FWD site will soon be redesigned, largely to make it simpler and more user-friendly: you'll be able to get your SIP credentials for free with one click, but that credential will die in 30 days unless you're a paid member. He admits that'll allow people to simply get a new one for free every 30 days—and he notes that, similarly,
[asterisk-users] Linksys 3102 with rfc2833 - NOT WORKING
I have two Sipura 3000 setup with Asterisk using dtmfmode=rfc2833 and everything is working perfectly. I setup another system using Linksys 3102 with Asterisk and I can not get RFC 2833 to work. They have identical setting when it comes to Audio Configuration except Linksys 3102 has an additional setting: DTMF Tx Mode: which Sipura 3K doesn't have. Sipura 3K works; Linksys 3102 DOES NOT I've tried all kinds of configuration under DTMF: AVT, Auto; Strick, Normal I'm running the latest firmware: 5.1.7(GW) Does anybody have a Linksys 3102 working with Asterisk and RFC2833? If so could someone please share the configuration (Private email is OK). -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding G.729 files
On Tue, Sep 23, 2008 at 4:44 AM, Alex Balashov [EMAIL PROTECTED] wrote: SOX will do it if you install its G.729 format library. As far as converting a group of files, that's what scripting is for, i.e. for FILE in `find . -type f -name '*.g729'`; do NFILE=$(echo $FILE | sed 's/\.g729/\.wav/g') sox [some args] $FILE ... $NFILE ... done Where can one find this? Is it legal? I don't want to get into all of that... For a known good way to convert (not to mention, %100 legal) you can just use Asterisk. Look at res_convert. Just make sure you have the G729 codec loaded. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive
On Tue, 23 Sep 2008, Steve Totaro wrote: FYI It looks like FWD is looking for value added service ideas for free as a volunteer. I got this too - looks like a bit of a mass mailling! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seemingly easy question: NPA/NXX
- Ira [EMAIL PROTECTED] wrote: At 09:29 AM 9/22/2008, you wrote: ... except in some countries, the phone numbers vary in length in the same city. Say in Hamburg, Germany, your number can be as short as 5 digits or as long as 10. You really have no way of knowing. The unanswered part of that, is this? Can 5 digit number, say, 12345, be the beginning part of a 10 digit number, say, 1234567890? And the answer is: do not confuse E.164 addresses with dialling patterns, grasshopper. Cheers, -- jr '1-888-MITSU2008' a -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seemingly easy question: NPA/NXX
- Karl Fife [EMAIL PROTECTED] wrote: Theory 1 Is it all done with timeouts, but they're CONDITIONAL timeouts. i.e. give a LONG timeout if the number: -did not start with a 1 and is still shorter than 7 digits, -started with a 1 and is still shorter than 11 digits -started with a 011 and is shorter than the theoretical international minimum lenght Theory 2 As you know, a few years ago the 2nd digit of the NPA was always 1 or 0. Therefore the switch could easily determine(without the leading 1) if your first three digits were an NPA or just an NXX (exchange). They were nationally unambiguous. Now that's no longer true. STILL, it could be possible to consider all known valid NPA's and exchanges so they can determine via context what you're trying to do, and thereby optimize the dialing experience? Can anyone speak to this? I would very much appreciate any knowledgable input. Well, my input is knowledgeable, though not authoritative. Yes, each NANP switch actually does have a routing table loaded locally (they call them translations) that tells it where to route calls for each and every valid NPA-NXX in the NANP, and this could be used to authenticate the first 3/6 digits of 7/10/11 digit dialled numbers for intra-NANP calls, and in fact, I would bet that you're correct that that's how they accomplish it. I have never actually seen live switch code on this, but I think I could locate some people who have -- but yes, you'll play hell duplicating it exactly on something with as small a brain as an ATA. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive
Gordon Henderson wrote: On Tue, 23 Sep 2008, Steve Totaro wrote: FYI It looks like FWD is looking for value added service ideas for free as a volunteer. I got this too - looks like a bit of a mass mailling! And me! And I haven't visited their site, or connected to their servers as IAX2 never worked, for well over a year either. Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
Hi Erik, once we used grandstream ATAs but now we are using linksys models: it has better design (look is important too for customers) and has 2 ports for two analog devices. We tested it with PRI and BRI lines and it seems working fine! Giorgio Incantalupo Erik Haider Forsen wrote: Hi Giorgio, Thanks for your answer. Your setup is exactly what we're thinking of. We have 1100 DID's, so that shouldn't be a problem at all. Which ATA box are you using? Erik On Sep 23, 2008, at 2:06 PM, Giorgio Incantalupo wrote: Hi Olivier, We DO NOT use faxdetect because it does not work properly. That's why we link a PRI DID to it, so when people call that DID the fax machine gets direct fax data without passing thru faxdetection. Giorgio Incantalupo. Olivier wrote: 2008/9/23 Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi Erik, we use an ATA device connected to the fax machine. If you want to receive faxes, since Asterisk fax detection is not reliable Hi, Which fax detection did you used, then ? , use one DID to link it directly to the ATA: you lose a number but you gain a fully-working fax! Giorgio Incantalupo. Erik Haider Forsen wrote: Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive
On Tue, Sep 23, 2008 at 04:48:36PM +0100, Gordon Henderson wrote: On Tue, 23 Sep 2008, Steve Totaro wrote: FYI It looks like FWD is looking for value added service ideas for free as a volunteer. I got this too - looks like a bit of a mass mailling! Mass-mailing to their (ex?-)customers/users. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting TE212p to NEC XenMaster
Hello list. Looking around I can't seem to find answers to what I am after, so here goes: I have an NEC Xen Master system (3 unit) basically maxed out. I want to connect a spare E1 card to the back of an existing Asterisk system terminating on a TE212P so I can divert out VOIP calls and eventually migrate over to Asterisk using the NEC system for handling the existing digital handsets only. So two questions: 1) Has anyone got any experience in connecting up this type of NEC system to a TE212 to route calls from the NEC system TO the Asterisk box? Any feedback / gotchas? 2) In this scenario, is the echo canceller needed? Or do I only really need it if I plan to route the call traffic out into another E1 circuit and to a telco? Thanks. Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro [EMAIL PROTECTED] wrote: ATAs work OK I guess, just make sure to use a loss less codec such as ULAW. Since the OP stated he is using E1 lines then he should probably be using alaw instead. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extension definition
Hi all, I need please the exact extension definition under extensions.conf that accepts any call coming from an appropriate username and Ip address...This mean that the authentication should be done on username and IP address Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Short question: CPU hardware requirements for Asterisk
Dear all, just a short question: What is the best CPU hardware requirements (CPU, memory, hard drive) to install Asterisk with SIP/RTP protocol for 100-150 users, and routing the RTP traffic by itself (no direct RTP traffic client-to-client) Special thanks Alejandro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4.21.2] Checking that already off-hook?
On Tue, 23 Sep 2008 12:29:22 +0200, Vincent
[EMAIL PROTECTED] wrote:
Isn't there a way to check the status an FXO card is in?
Apparently, it's OK to call Answer() even if the channel is already
open:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Answer
So I guess I can simplify things this way:
[my-ivr]
HELLO=false
exten = s,1,GotoIf($[${LEN(${CALLERID(num)})} = 0]?nocid,1:cid,1)
exten = nocid,1,Set(HELLO=true);
exten = nocid,n,Answer()
exten = nocid,n,Playback(my_sound_files/hello)
exten = nocid,n,Read(CALLERID(num),my_sound_files/no_cid,10)
exten = nocid,n,GotoIf($[${LEN(${CALLERID(num)})} 10]?cid,1)
exten = nocid,n,Hangup()
;If number in DB, rewrite CID name on the fly
exten =
cid,1,AGI(check_cid.phpcli|${CALLERID(num)}|${CALLERID(name)})
exten = cid,n,Goto(main_menu,s,1)
[main_menu]
;OK to call Answer() even if line already off-hook
exten = s,1,Answer()
exten = s,n,ExecIf($[${HELLO} = true],Playback,my_sound_files/hello)
exten = s,n,Background(my_sound_files/main_menu)
exten = s,n,WaitExten(5)
exten = s,n,Hangup()
Thank you.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive
I was interested in participating but receive no reply at all about how to be on the call. Nor was Jeff Pulver on Facebook or responding to Twitter. And I paid for my membership thinking that I'd give it a year and see what happened. Michael On Tue, 23 Sep 2008 11:16:09 -0400, Steve Totaro wrote: FYI It looks like FWD is looking for value added service ideas for free as a volunteer. I think it will fail but we shall see. I really don't get the nerve of them (Free World Dialup has changed it's name to FWD) to ask for free ideas and development on a non-free service. Maybe if they can come up with a killer app and people will adopt it, then it might work, but then again, people still cling to their analog FAX machines Thanks, Steve Totaro -- Forwarded message -- From: Daniel Berninger [EMAIL PROTECTED] Date: Tue, Sep 23, 2008 at 10:39 AM Subject: more on Free World Dialup groups and FWDLive To: [EMAIL PROTECTED] Hello, We are looking for group leaders and topic ideas for the FWD voice analog of Yahoo!Groups - FWDLive. The exact approach to FWDLive remains a work in progress. We know FWDLive should offer SIP enabled group conversations along the lines of an open protocol version of Talkshoe. We may end up limiting the size and access to groups to avoid the sort of disruptive participants that led to the demise of Skypecasts. A prototype of process for creating groups will get posted to FWDWiki: 1) pick a topic and write short summary 2) pick a time to run the call, post to the schedule, request conference code 3) dial into the group at the appointed time Jeff Pulver will host a call today at 2:00 ET to discuss FWDLive topic ideas. Reply to this note if your are interested in joining the call with Jeff or volunteering as a group leader. I also attached a VoIP Planet article below that provides more details on why FWD moved to paid membership. Best regards, Dan ... Daniel Berninger CEO, FWD fwd: 12908 v: +1.202.250.3838 e: [EMAIL PROTECTED] w: www.freeworlddialup.com http://www.voipplanet.com/news/article.php/3767266 Free World Dialup No Longer Free August 22, 2008 By Jeff Goldman FWD, formerly known as Free World Dialup, will next month start charging a mandatory subscription fee of $30 per year, as part of a larger plan to reinvent itself as what the company calls a 'Communication ISP.' This follows FWD's introduction a year ago of an optional $30-a-year membership plan. According to FWD CEO Daniel Berninger, the mandatory fee was simply a logical next step. The voluntary one gave us the confidence to do the required one... it was pretty successful, so what we ended up figuring out over the year was that we wanted to be able to fund ourselves enough so that we wouldn't have to do any kind of PSTN funding, like selling DIDs, he says. And that, Berninger says, is really the point. After a decade, VoIP hasn't reached its potentialit basically is an on-ramp to the telephone network, and doesn't do anything else, he says. People have experimented with things, but for the most part, all the revenue models of [companies like] Skype and JAJAH... have something to do with extracting money based on usage charges and giving people access to the telephone network. Instead, Berninger wants to turn FWD into a Communication ISP, an idea he introduced in a blog post earlier this month in which he argued that Interconnection with the telephone network shuts out the possibility of creativity... Content is limited to those uses justified in the context of the per minute cost of telephone service. And so the Communication ISP is intended to be a pure SIP offering, free of the PSTN and its inherent restrictions. For your regular ISP, you pay them a monthly fee and they attach your computer to the Internet... we want to be the same thing, in that you buy a communication device, a SIP VoIP device, and you go to a Communication ISP and get the thing on the Internet... and from there, you build applications and create new value, he says. So we're thinking about this like an entire ecosystem. To compete with the dominance of the PSTN, Berninger says, VoIP needs to differentiate itself better, not only with things like video and wideband audio, but also with a whole new range of as-yet-unknown applications. The hard part of the argument is this bootstrap problemin other words, how do we get from where we are, not knowing what the applications are and not having anybody with capable devices, to scale? he says. The parallel, of course, would be the early days of the Internet. When it started, there was a very small audience and very limited content, but it did have global termination for the same price... and it created the virtuous cycle of content attracting more audience and audience attracting more contentand the next thing you know, the thing's growing tenfold a year, he says. To begin with, Berninger says, the FWD site will soon be
Re: [asterisk-users] extension definition
On Tue, 23 Sep 2008, michel freiha wrote: I need please the exact extension definition under extensions.conf that accepts any call coming from an appropriate username and Ip address...This mean that the authentication should be done on username and IP address Guessing based on the information provided... Authentication is configured by iax.conf or sip.conf. Search for details on voip-info.org. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Short question: CPU hardware requirements for Asterisk
On Tue, 23 Sep 2008, Alejandro Cabrera Obed wrote: Dear all, just a short question: What is the best CPU hardware requirements (CPU, memory, hard drive) to install Asterisk with SIP/RTP protocol for 100-150 users, and routing the RTP traffic by itself (no direct RTP traffic client-to-client) A short question does not imply a short answer :) Best depends on your Clinton-esq definition of whatever best means to you. CPU - any reasonably modern, mainstream processor -- assuming you are not transcoding. RAM - The Asterisk process will consume about 100mb. Disk - Irrelevant to processing calls. You can build a CentOS based system on less than 4gb. Astlinux can do it on the head of a pin. If you want more specific answers you need to spend some time developing more specific questions :) Searching about on voip-info.org for dimensioning may help. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 or 1.6
I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo portage but I think this version has a problem with RFC2833 DTMF signaling and I don't think there will be any newer version available anytime soon on portage. I need stable version, I'm using Asterisk mostly with ATA adapter (Linksys and Sipura); should I go to 1.6 or 1.4? -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extension definition
This is done in sip.conf, iax.conf, etc, not in extensions.conf. By the time a call gets to extensions.conf it must already be authenticated. Assume the username is robertdobbs and the ip is 209.17.71.61 In sip.conf you would have something like this: [robertdobbs] deny=0.0.0.0/0 permit=209.17.71.61 rest of the options here michel freiha wrote: Hi all, I need please the exact extension definition under extensions.conf that accepts any call coming from an appropriate username and Ip address...This mean that the authentication should be done on username and IP address Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6
On Tue, 23 Sep 2008, Joseph wrote: I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo portage but I think this version has a problem with RFC2833 DTMF signaling and I don't think there will be any newer version available anytime soon on portage. I need stable version, I'm using Asterisk mostly with ATA adapter (Linksys and Sipura); should I go to 1.6 or 1.4? Compile up 1.2.30 yourself :) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and prepaid billing + Radius
Yes it answer and big thanks. I have another question (which might be not related alot to AGI) if u can help me: If Asterisk support Radius, so we can build Prepaid Billing with Radius to communicate via Radius as standard communication method? Regards Bilal --- On Tue, 9/23/08, Benjamin Jacob [EMAIL PROTECTED] wrote: From: Benjamin Jacob [EMAIL PROTECTED] Subject: Re: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com, [EMAIL PROTECTED] Date: Tuesday, September 23, 2008, 6:39 AM Hi Bilal, Yes it is definitely possible. And I've done it myself for a couple of our clients. Does that answer your two questions? cheers - Ben. --- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote: From: bilal ghayyad [EMAIL PROTECTED] Subject: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com Date: Tuesday, September 23, 2008, 9:52 AM Hi All; Did anyone do an prepaid billing application via AGI? I would like to know if that is possible. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and prepaid billing + Radius
You could. bilal ghayyad wrote: Yes it answer and big thanks. I have another question (which might be not related alot to AGI) if u can help me: If Asterisk support Radius, so we can build Prepaid Billing with Radius to communicate via Radius as standard communication method? Regards Bilal --- On Tue, 9/23/08, Benjamin Jacob [EMAIL PROTECTED] wrote: From: Benjamin Jacob [EMAIL PROTECTED] Subject: Re: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com, [EMAIL PROTECTED] Date: Tuesday, September 23, 2008, 6:39 AM Hi Bilal, Yes it is definitely possible. And I've done it myself for a couple of our clients. Does that answer your two questions? cheers - Ben. --- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote: From: bilal ghayyad [EMAIL PROTECTED] Subject: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com Date: Tuesday, September 23, 2008, 9:52 AM Hi All; Did anyone do an prepaid billing application via AGI? I would like to know if that is possible. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: more on Free World Dialup groups and FWDLive
Apparently this call was deffered...but noone was told. I tried to get on it for 30 minutes, and tried to contact various people at FWD. Dan Behrninger evetually responded to say it was to be rescheduled. Michael On Tue, 23 Sep 2008 13:24:07 -0500, Michael Graves wrote: I was interested in participating but receive no reply at all about how to be on the call. Nor was Jeff Pulver on Facebook or responding to Twitter. And I paid for my membership thinking that I'd give it a year and see what happened. Michael On Tue, 23 Sep 2008 11:16:09 -0400, Steve Totaro wrote: FYI It looks like FWD is looking for value added service ideas for free as a volunteer. I think it will fail but we shall see. I really don't get the nerve of them (Free World Dialup has changed it's name to FWD) to ask for free ideas and development on a non-free service. Maybe if they can come up with a killer app and people will adopt it, then it might work, but then again, people still cling to their analog FAX machines Thanks, Steve Totaro -- Forwarded message -- From: Daniel Berninger [EMAIL PROTECTED] Date: Tue, Sep 23, 2008 at 10:39 AM Subject: more on Free World Dialup groups and FWDLive To: [EMAIL PROTECTED] Hello, We are looking for group leaders and topic ideas for the FWD voice analog of Yahoo!Groups - FWDLive. The exact approach to FWDLive remains a work in progress. We know FWDLive should offer SIP enabled group conversations along the lines of an open protocol version of Talkshoe. We may end up limiting the size and access to groups to avoid the sort of disruptive participants that led to the demise of Skypecasts. A prototype of process for creating groups will get posted to FWDWiki: 1) pick a topic and write short summary 2) pick a time to run the call, post to the schedule, request conference code 3) dial into the group at the appointed time Jeff Pulver will host a call today at 2:00 ET to discuss FWDLive topic ideas. Reply to this note if your are interested in joining the call with Jeff or volunteering as a group leader. I also attached a VoIP Planet article below that provides more details on why FWD moved to paid membership. Best regards, Dan ... Daniel Berninger CEO, FWD fwd: 12908 v: +1.202.250.3838 e: [EMAIL PROTECTED] w: www.freeworlddialup.com http://www.voipplanet.com/news/article.php/3767266 Free World Dialup No Longer Free August 22, 2008 By Jeff Goldman FWD, formerly known as Free World Dialup, will next month start charging a mandatory subscription fee of $30 per year, as part of a larger plan to reinvent itself as what the company calls a 'Communication ISP.' This follows FWD's introduction a year ago of an optional $30-a-year membership plan. According to FWD CEO Daniel Berninger, the mandatory fee was simply a logical next step. The voluntary one gave us the confidence to do the required one... it was pretty successful, so what we ended up figuring out over the year was that we wanted to be able to fund ourselves enough so that we wouldn't have to do any kind of PSTN funding, like selling DIDs, he says. And that, Berninger says, is really the point. After a decade, VoIP hasn't reached its potentialit basically is an on-ramp to the telephone network, and doesn't do anything else, he says. People have experimented with things, but for the most part, all the revenue models of [companies like] Skype and JAJAH... have something to do with extracting money based on usage charges and giving people access to the telephone network. Instead, Berninger wants to turn FWD into a Communication ISP, an idea he introduced in a blog post earlier this month in which he argued that Interconnection with the telephone network shuts out the possibility of creativity... Content is limited to those uses justified in the context of the per minute cost of telephone service. And so the Communication ISP is intended to be a pure SIP offering, free of the PSTN and its inherent restrictions. For your regular ISP, you pay them a monthly fee and they attach your computer to the Internet... we want to be the same thing, in that you buy a communication device, a SIP VoIP device, and you go to a Communication ISP and get the thing on the Internet... and from there, you build applications and create new value, he says. So we're thinking about this like an entire ecosystem. To compete with the dominance of the PSTN, Berninger says, VoIP needs to differentiate itself better, not only with things like video and wideband audio, but also with a whole new range of as-yet-unknown applications. The hard part of the argument is this bootstrap problemin other words, how do we get from where we are, not knowing what the applications are and not having anybody with capable devices, to scale? he says. The parallel, of course, would be the early days of the Internet. When it started, there was a very small audience and very limited content, but it did have global
Re: [asterisk-users] AGI and prepaid billing + Radius
Hi Bilal, Asterisk's RADIUS support is limited to CDRs, that is, the last A in AAA (Accounting). As for Authentication and Authorization, Asterisk integrates very well with PortaOne's billing systems (PortaBilling + PortaSIP), if you use their PERL RADIUS client : http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth I guess if you tweak that RADIUS client a bit, you can make it work with any RADIUS based billing system. Cheers, Philippe On Tue, Sep 23, 2008 at 10:35 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Yes it answer and big thanks. I have another question (which might be not related alot to AGI) if u can help me: If Asterisk support Radius, so we can build Prepaid Billing with Radius to communicate via Radius as standard communication method? Regards Bilal --- On Tue, 9/23/08, Benjamin Jacob [EMAIL PROTECTED] wrote: From: Benjamin Jacob [EMAIL PROTECTED] Subject: Re: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com, [EMAIL PROTECTED] Date: Tuesday, September 23, 2008, 6:39 AM Hi Bilal, Yes it is definitely possible. And I've done it myself for a couple of our clients. Does that answer your two questions? cheers - Ben. --- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote: From: bilal ghayyad [EMAIL PROTECTED] Subject: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com Date: Tuesday, September 23, 2008, 9:52 AM Hi All; Did anyone do an prepaid billing application via AGI? I would like to know if that is possible. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Philippe Sultan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No route to destination error
Hi, I'm sorry for cross-posting this (from http://forums.digium.com/viewtopic.php?t=64280), but I havn't got any replies in the forum.. When I dial out from my Asterisk 1.4.19 installation on Debian (three SIP hardphones on a LAN, and an IAX2 connection over DSL to a commercial trunk) I get this error on the console: -- Executing [EMAIL PROTECTED]:1] Set(SIP/21-081ceea8, CALLERID(all)= 88821268) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/21-081ceea8, IAX2/88821268/40618405|30|r) in new stack [Sep 11 12:05:58] WARNING[7098]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Congestion(SIP/21-081ceea8, ) in new stack == Spawn extension (default, 40618405, 3) exited non-zero on 'SIP/21-081ceea8' I can't see any traffic on the wire using ngrep, and the registry looks good: filserver*CLI iax2 show registry Host dnsmgr Username Perceived Refresh State 85.nnn.nnn.83:4569 N 88821268 85.nnn.nn.197:1 60 Registered 85.nnn.nnn.82:4569 N 88821268 85.nnn.nn.197:10002 60 Registered I can see traffic with ngrep while registering, and every 60 seconds after that. That no route to destination error is causing my hair to thin, and my trunk provider tells me that it's usually something else, and that the errormessage is not that descriptive. What can I do to get more/better debugging info? I can't figure out what's wrong. Thanks! - Martin ( my iax.conf and extensions.conf on http://pastebin.com/mb0020bd ) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and prepaid billing + Radius
Yes, of course you can. We have used Perl and Authen::Radius in the past to create AGI calling card scripts to do AAA against RADIUS servers. Not only that, but we used it for routing the outgoing calls also in many cases. Best regards, Vlasis Hatzistavrou. bilal ghayyad wrote: Yes it answer and big thanks. I have another question (which might be not related alot to AGI) if u can help me: If Asterisk support Radius, so we can build Prepaid Billing with Radius to communicate via Radius as standard communication method? Regards Bilal --- On Tue, 9/23/08, Benjamin Jacob [EMAIL PROTECTED] wrote: From: Benjamin Jacob [EMAIL PROTECTED] Subject: Re: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com, [EMAIL PROTECTED] Date: Tuesday, September 23, 2008, 6:39 AM Hi Bilal, Yes it is definitely possible. And I've done it myself for a couple of our clients. Does that answer your two questions? cheers - Ben. --- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote: From: bilal ghayyad [EMAIL PROTECTED] Subject: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com Date: Tuesday, September 23, 2008, 9:52 AM Hi All; Did anyone do an prepaid billing application via AGI? I would like to know if that is possible. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and prepaid billing + Radius
Dear Philippe; Thanks a lot for ur kindly answer. How can I use the Radius with CDR (Accounting)? About PortaOne's billing systems: Do u mean I can use the PortaOne's billing systems Radius client (to be fixed at Asterisk side), and customize this client to be used with any RADIUS based billing system? Your kindly help is high appreciated. Regards Bilal --- On Tue, 9/23/08, Philippe Sultan [EMAIL PROTECTED] wrote: From: Philippe Sultan [EMAIL PROTECTED] Subject: Re: [asterisk-users] AGI and prepaid billing + Radius To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, September 23, 2008, 4:54 PM Hi Bilal, Asterisk's RADIUS support is limited to CDRs, that is, the last A in AAA (Accounting). As for Authentication and Authorization, Asterisk integrates very well with PortaOne's billing systems (PortaBilling + PortaSIP), if you use their PERL RADIUS client : http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth I guess if you tweak that RADIUS client a bit, you can make it work with any RADIUS based billing system. Cheers, Philippe On Tue, Sep 23, 2008 at 10:35 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Yes it answer and big thanks. I have another question (which might be not related alot to AGI) if u can help me: If Asterisk support Radius, so we can build Prepaid Billing with Radius to communicate via Radius as standard communication method? Regards Bilal --- On Tue, 9/23/08, Benjamin Jacob [EMAIL PROTECTED] wrote: From: Benjamin Jacob [EMAIL PROTECTED] Subject: Re: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com, [EMAIL PROTECTED] Date: Tuesday, September 23, 2008, 6:39 AM Hi Bilal, Yes it is definitely possible. And I've done it myself for a couple of our clients. Does that answer your two questions? cheers - Ben. --- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote: From: bilal ghayyad [EMAIL PROTECTED] Subject: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com Date: Tuesday, September 23, 2008, 9:52 AM Hi All; Did anyone do an prepaid billing application via AGI? I would like to know if that is possible. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Philippe Sultan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A2Billing Callback Hangup after/about 20 sec!
Hi! I am posting a2billing issue here in asterisk list because some one might have faced same issue with a2billing callback. My Callback problem has been already posted on a2billing forum which I am facing on my system. Please have a look on this thread: http://forum.asterisk2billing.org/viewtopic.php?t=3093 I am using CID-Callback, when you call the access number, it calls back and ask for the destination number, as you enter the destination number the call will hangup about 20 - 24 seconds, as you are entering the destination number. I am using Asterisk 1.4, A2billing 1.3.3 on Centos 5.2 every thing is fine except callback hangup issue. by digging Google and forums for same issue, I got answered to change the carrier and i have tried with six different carrier, and it works very rare among one out of ten calls. Can any one suggest how to fix it. I have emailed a2billing support also for resolving issue but not yet got any reply. Thanks Regards Zulqarnain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No route to destination error
Martin Seebach schrieb: When I dial out from my Asterisk 1.4.19 installation on Debian (three SIP hardphones on a LAN, and an IAX2 connection over DSL to a commercial trunk) I get this error on the console: -- Executing [EMAIL PROTECTED]:2] Dial(SIP/21-081ceea8, IAX2/88821268/40618405|30|r) in new stack [Sep 11 12:05:58] WARNING[7098]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Congestion(SIP/21-081ceea8, ) in new stack I can't see any traffic on the wire using ngrep, and the registry looks good: filserver*CLI iax2 show registry Host dnsmgr Username Perceived Refresh State 85.nnn.nnn.83:4569 N 88821268 85.nnn.nn.197:1 60 Registered 85.nnn.nnn.82:4569 N 88821268 85.nnn.nn.197:10002 60 Registered I can see traffic with ngrep while registering, and every 60 seconds after that. Maybe something is broken in recent versions of chan_iax2.c? http://lists.digium.com/pipermail/asterisk-users/2008-September/218560.html Not the same issue though. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extension definition
Hello Eric, i didwhat you asked me to do but i'm getting Notfound sip message when trying to register regrads On Tue, Sep 23, 2008 at 9:56 PM, Eric ManxPower Wieling [EMAIL PROTECTED]wrote: This is done in sip.conf, iax.conf, etc, not in extensions.conf. By the time a call gets to extensions.conf it must already be authenticated. Assume the username is robertdobbs and the ip is 209.17.71.61 In sip.conf you would have something like this: [robertdobbs] deny=0.0.0.0/0 permit=209.17.71.61 rest of the options here michel freiha wrote: Hi all, I need please the exact extension definition under extensions.conf that accepts any call coming from an appropriate username and Ip address...This mean that the authentication should be done on username and IP address Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to hangup a channel immediately so that it doesn't get charged on cell phone
Thanks for your reply. Yes, it came from another context where it was first
answered. I put there s,1,Hangup and then ran other priorities. I hope this
will fix my problem, but not sure yet.
Zeeshan
On Tue, Sep 23, 2008 at 3:38 AM, Tony Mountifield
[EMAIL PROTECTED]wrote:
In article [EMAIL PROTECTED],
Zeeshan Zakaria [EMAIL PROTECTED] wrote:
On my call back system, I have the script as follows:
[calback]
exten = s,1,NoOp(* STARTING CALLCHEAP\'S CALLBACK SYSTEM *)
exten = s,n,Set(CALL=${CALLERID(number)})
exten = s,n,Set(DESTINATION=myCallback.2000.1)
exten = s,n,Set(SLEEP=5)
exten = s,n,System(/var/lib/asterisk/bin/callback ${CALL} ${DESTINATION}
${SLEEP} )
exten = s,n,Hangup
The idea behind this system is that the script picks up the call, notes
down
the caller's number, and hangs it immediately. Then the caller gets a
call
back.
But what is happening is that cell phone callers are still being charged
for
calling into this callback context.
How can I avoid this? I want cell phone users to not get charged for the
call back.
How does the incoming call get to calback,s,1 ? Is there another part of
the dialplan that receives the call and then jumps to here? If so, you
need to make sure that it doesn't call Answer(), nor any application that
might do an implicit answer.
Otherwise, please give more details about how the calls are delivered to
your system, and what you do with them right from the beginning.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Zeeshan A Zakaria
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to send indicating call privacy using P-Asserted-Identity?
Hi, I know how to use indicating P-Asserted-Identity, but the SIP trunk provider requires to send call privacy using P-Asserted-Identity or Remote-Party-Id header. What I am doing is exten = _.,n,SipAddHeader(P-Asserted-Identity: name sip:[EMAIL PROTECTED]) The provider gets this as anonymous and can't flag the call as private. On asking them again, they say send us Invites that either contain Remote-Party-ID or P-Asserted-Identity with the correct headers flagged. Now I don't know what exactly this means and how do I do this. They are SIP trunk providers but don't deal is Asterisk. Any ideas? -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6
Joseph wrote: I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo portage but I think this version has a problem with RFC2833 DTMF signaling and I don't think there will be any newer version available anytime soon on portage. I need stable version, I'm using Asterisk mostly with ATA adapter (Linksys and Sipura); should I go to 1.6 or 1.4? By saying you need a stable version, you've answered your own question. 1.6 has not yet been released and therefore should not be considered to be production ready yet. Upgrade to version 1.6 at your own risk only. Some people have reported 1.6 to be very stable, others (such as myself) are still having occasional problems with it. For me, these issues aren't a great concern since it's a home system, but at least for the moment, I wouldn't consider running 1.6 in a production environment. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to hangup a channel immediately so that it doesn't get charged on cell phone
Don't hangup just don't answer either. You could probably just start
at [callback], what does the other context do?
Thanks,
Steve Totaro
On Tue, Sep 23, 2008 at 6:54 PM, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Thanks for your reply. Yes, it came from another context where it was first
answered. I put there s,1,Hangup and then ran other priorities. I hope this
will fix my problem, but not sure yet.
Zeeshan
On Tue, Sep 23, 2008 at 3:38 AM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],
Zeeshan Zakaria [EMAIL PROTECTED] wrote:
On my call back system, I have the script as follows:
[calback]
exten = s,1,NoOp(* STARTING CALLCHEAP\'S CALLBACK SYSTEM *)
exten = s,n,Set(CALL=${CALLERID(number)})
exten = s,n,Set(DESTINATION=myCallback.2000.1)
exten = s,n,Set(SLEEP=5)
exten = s,n,System(/var/lib/asterisk/bin/callback ${CALL}
${DESTINATION}
${SLEEP} )
exten = s,n,Hangup
The idea behind this system is that the script picks up the call, notes
down
the caller's number, and hangs it immediately. Then the caller gets a
call
back.
But what is happening is that cell phone callers are still being charged
for
calling into this callback context.
How can I avoid this? I want cell phone users to not get charged for the
call back.
How does the incoming call get to calback,s,1 ? Is there another part of
the dialplan that receives the call and then jumps to here? If so, you
need to make sure that it doesn't call Answer(), nor any application that
might do an implicit answer.
Otherwise, please give more details about how the calls are delivered to
your system, and what you do with them right from the beginning.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Zeeshan A Zakaria
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to notify an event to every user
I have found that with the right diet, teargas is not necessary. That interesting to know. Maybe we should open a new thread on that and let everyone contribute ;-) I still think it's a valid idea - with the right lunch, you could guarantee that the office was empty (except for yourself) PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cellroute setup with asterisk
OK, here is how it is working so far: Cellroute has a 3G sim card in it. Its Phone port is connected to TDM400P FXO port, just next to my incoming BT line. Calling out works fine - just as with the BT line. On incoming calls there seems to be a problem with caller ID chan_zap.c:4155 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. Also does not get me to voicemail with error messages as follows: -- Zap/4-1 is ringing -- Nobody picked up in 15000 ms -- Hungup 'Zap/4-1' -- Executing [EMAIL PROTECTED]:4] Goto(Zap/2-1, s-NOANSWER|1) in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing [EMAIL PROTECTED]:1] VoiceMail(Zap/2-1, 10| u) in new stack -- Zap/2-1 Playing 'vm-theperson' (language 'en') == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'Zap/2-1' in macro 'stdexten' == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' However when I set up the incoming call to go directly to voicemail everything seems to work alright. Next things to do will be to create a voice menu and see how well DTMF tones are recognised. Then I plan to add Cellroute to LAN (for sending SMS) and to serial port (so that the default IP address can be changed and perhaps other useful things done as well). Robert On Wed, 2008-09-17 at 09:04 +0100, Roberts Klotins wrote: Hi there! Sorry, I should have started this as a separate thread. Here we go: I wonder if anyone has set up Cellroute or Cellroute 3G mobile network gateway (see http://www.gsmsave.com/acatalog/CellRoute-3G.pdf ) with asterisk. I am about to do that soon, therefore any experience would be highly appreciated. I understand that one could connect the PSTN port on it to a FXO port on a TDM400P card and that probably could take care of calling. I wonder how then is it possible to deal with SMS? Best wishes, Robert P.S. And of course I will be posting followups to inform how I am getting along with the setup. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to hangup a channel immediately so that it doesn't get charged on cell phone
The other context was running the AGI script to capture caller ID and then
sending it to this context. Now I've changed it. It hangs up right in the
first context. I can see SIP message 603 in SIP header now. So I guess now
the cell phone providers should see it as a not answered call.
Zeeshan
On Tue, Sep 23, 2008 at 7:19 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Don't hangup just don't answer either. You could probably just start
at [callback], what does the other context do?
Thanks,
Steve Totaro
On Tue, Sep 23, 2008 at 6:54 PM, Zeeshan Zakaria [EMAIL PROTECTED]
wrote:
Thanks for your reply. Yes, it came from another context where it was
first
answered. I put there s,1,Hangup and then ran other priorities. I hope
this
will fix my problem, but not sure yet.
Zeeshan
On Tue, Sep 23, 2008 at 3:38 AM, Tony Mountifield
[EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED]
,
Zeeshan Zakaria [EMAIL PROTECTED] wrote:
On my call back system, I have the script as follows:
[calback]
exten = s,1,NoOp(* STARTING CALLCHEAP\'S CALLBACK SYSTEM *)
exten = s,n,Set(CALL=${CALLERID(number)})
exten = s,n,Set(DESTINATION=myCallback.2000.1)
exten = s,n,Set(SLEEP=5)
exten = s,n,System(/var/lib/asterisk/bin/callback ${CALL}
${DESTINATION}
${SLEEP} )
exten = s,n,Hangup
The idea behind this system is that the script picks up the call,
notes
down
the caller's number, and hangs it immediately. Then the caller gets a
call
back.
But what is happening is that cell phone callers are still being
charged
for
calling into this callback context.
How can I avoid this? I want cell phone users to not get charged for
the
call back.
How does the incoming call get to calback,s,1 ? Is there another part of
the dialplan that receives the call and then jumps to here? If so, you
need to make sure that it doesn't call Answer(), nor any application
that
might do an implicit answer.
Otherwise, please give more details about how the calls are delivered to
your system, and what you do with them right from the beginning.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Zeeshan A Zakaria
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Zeeshan A Zakaria
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Short question: CPU hardware requirements for Asterisk
Steve Edwards wrote: On Tue, 23 Sep 2008, Alejandro Cabrera Obed wrote: Dear all, just a short question: What is the best CPU hardware requirements (CPU, memory, hard drive) to install Asterisk with SIP/RTP protocol for 100-150 users, and routing the RTP traffic by itself (no direct RTP traffic client-to-client) Hi Maybe below document will help you with an idea what is @[EMAIL PROTECTED] http://www.bicomsystems.com/files/whitepapers/report-officeBOX-testing.pdf Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk mysql CDR
hi,
i'm using this macro to dial an extension and forward to a mobile if
unavailable,busy or noanswer
exten = 100,1,Macro(dial-ext|SIP/${EXTEN}|vm-100|moh-100)
exten = 100,2,Goto(100-${DIALSTATUS}|1)
exten = 100-BUSY,1,Macro(dialout-local-mobile|91234567)
exten = 100-BUSY,2,Voicemail([EMAIL PROTECTED]|u)
exten = 100-CONGESTION,1,Macro(dialout-local-mobile|91234567)
exten = 100-CONGESTION,2,Voicemail([EMAIL PROTECTED]|u)
exten = 100-NOANSWER,1,Macro(dialout-local-mobile|91234567)
exten = 100-NOANSWER,2,Voicemail([EMAIL PROTECTED]|u)
my prob is on the CDR, from extension 500 i called 100, 100 is not
online so it should forward it to my mobile
but on the cdr it shows like this:
FromTo
500 100-CHANUNAVAIL
should it be like
FromTo
500 91234567
or
FromTo
100 91234567
any idea how to fix those?
regards,
nhadie
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
