[asterisk-users] IAX2 Load test
Hello everybody. I have been running some load tests on IAX2, as we are finding out our future hardware investments. Here is the setup: We have three virtual machines. A: running SIPP and Asterisk 1.6.2.7 B: running Asterisk 1.6.2.7 C: running Asterisk 1.6.2.7. All of the Asterisks have been compiled with as little modules and channel drivers as possible, etc. only SIP, Local and IAX2, and about 4 dialplan functions. They are connected as this. A - IAX2 - B - SIP - C On the A machine, SIPP makes SIP calls to the asterisk, which then dials server B through IAX, which then in turn dials server C through SIP. The scenario is to have machine B as a IAX2 - SIP converter. Asterisk on machine A and B dies around 185 simultaneous channels, consuming all CPU on the machines, until they eventually crash. Machine C, which is only running SIP, consumes around 30% CPU at the 185 channel mark. This result gutted me somewhat, as 185 channel is a really low figure. And 185 channels is where it crashed, so 160 channels would probably be a safer estimate per machine. When using only SIP we have been able to run over 860 calls on a single Asterisk 1.6, and the factor that stopped using more channels, was our Cisco PIX506 firewall crashing. Ive read several places, that IAX2 scales really horribly, and having confirmed that, I am wondering if anybody has a solution for this. My own idea was to develop a IAX2 - SIP procotol converter. Ive worked somewhat with the IAX2 protocol in code, and it should absolutely be possible, but unfortunately, i do not have the time for such a project. Any other ideas? Med venlig hilsen/Kind Regards Jon Leren Schøpzinsky -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 T.38 generation towards a Cisco voice router
Hello List. We are having some problems using t.38 together with a Cisco voice router at one of our providers end. We are using the new digium asterisk fax module to generate the fax, and when we use together with our internal Audiocodes Mediant 2000 gateways, we have no issues what so ever, and the faxes go right through. When we send faxes to our other provider, who has cisco hardware at their end, we get this error: -- Channel 'SIP/xxx.xx.se-08aaf470' fax session '0', [ 006.607923 ], STAT_EVT_HW_CLOSE st: WT_HW_CLSrt: WCLSNCLS -- Channel 'SIP/xxx.xx.se-08aaf470' fax session '0', [ 006.608118 ], STAT_SES_COMPLETE -- Channel 'SIP/xxx.xx.se-08aaf470' fax session '0' is complete, result: 'SUCCESS' (FAX_NO_FAX), error: 'CANCELED', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:33] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:33] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:33] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short I can see that asterisk discards all RTP T.38 packets sent from the provider, which the error message also indicates. Is there a known problem, connecting to cisco hardware using t.38 in Asterisk 1.6? or does anybody know of a patch that fixes this problem? I can see that in the end of the T.38 packet, cisco adds 4 zero fields, which are not in the packets that Asterisk sends. Is this some weird we-are-cisco-and-therefore-decide-how-the-packets-should-look? Kind Regards Jon Leren Schøpzinsky Systems Architect Firstcom A/S Bådehavnsgade 2C, 2. 2450 København SV Web: http://www.firstcom.dk http://www.firstcom.dk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 T.38 generation towards a Ciscovoice router
I used wireshark to debug the problem, and I can see that the cisco equipment is correctly sending t.38 packets to asterisk, and the whole re-invite process is successful. The problem is, that Asterisk discards the t.38 packets with the error message I sent, and therefore the T.38 session never gets underway. Asterisk is stuck on the same SEQ id, as it never receives anything from the cisco. Ive also checked that this isn't a network issue. The packets are coming through, asterisk just throws them away with the error message I described. Med venlig hilsen/Kind Regards Jon Leren Schøpzinsky Systems Architect Firstcom A/S Bådehavnsgade 2C, 2. 2450 København SV Web: http://www.firstcom.dk -Oprindelig meddelelse- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af David Backeberg Sendt: 13. maj 2009 14:12 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Asterisk 1.6 T.38 generation towards a Ciscovoice router On Wed, May 13, 2009 at 3:30 AM, Jon Schøpzinsky j...@firstcom.dk wrote: We are having some problems using t.38 together with a Cisco voice router at one of our providers end. We are using the new digium asterisk fax module to generate the fax, and when we use together with our internal Audiocodes Mediant 2000 gateways, we have no issues what so ever, and the faxes go right through. I can see that asterisk discards all RTP T.38 packets sent from the provider, which the error message also indicates. Is there a known problem, connecting to cisco hardware using t.38 in Asterisk 1.6? or does anybody know of a patch that fixes this problem? I doubt that there is a known problem, as I'm using Cisco with asterisk and T.38 and having success. Do you have full control over the Cisco gear? Please post the dialpeer info from the Cisco gear and I'll take a look at it. You can also go back through the archives for similar posts because we've discussed this a few times in the last few months. Among other things, I saw that your fax tried to transmit at 2400bps. The gear should be able to support 9600. So that's already fishy. What happens if you try to send a 'normal' audio fax over voip through that gear? Some things you should know: * do not compress voip faxes. Faxes are already compressed. If you try to use a compression codec you'll wreck the fax. * on the cisco dialpeer be darn sure that you've turned off vad * for sip on asterisk, you need to enable reinvite, and you also need to configure a t38pt_udptl = yes entry in your sip.conf, but you probably already to that right if you were T.38-ing to the other gear. Are you sure you weren't just sending a normal audio fax to the other gear? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message was scanned and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap Call deflection on PRI
Hello List. I have created a patch some time ago, to use the ISDN feature call deflection or partial rerouting, as it Is also known, to make proper call transfers on PRI, without using an extra channel for the outgoing call. Back then I ported the function zapCD from bristuff, to a normal zaptel, and it worked as it should. I am looking at this again, and was wondering if anybody has made a similar function for zaptel, since this is a very usefull feature, or if I should make a new patch myself. Kind Regards Jon Leren Schøpzinsky ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Call deflection on PRI
Actually this isnt the same as Two B-Channel transfer. This is done by sending a FACILITY message to the ISDN, which in hand then disconnects the call and sends it to the number provided in the call deflection message. All b-channels are closed when the FACILITY message is sent. You can also send a deflection reason code, such as Busy and Unavailable. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: 27. februar 2008 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Call deflection on PRI On Wed, 2008-02-27 at 08:50 -0500, Steve Totaro wrote: I recall that this is now part of Asterisk (1.4 or 1.6 or both). It really is a great feature rather than using two channels in trunk to trunk transfer. This is often called a Two B-Channel Transfer, or TBCT. As long as your PRI provider has this service enabled on your PRI, recent versions of libpri, zaptel, and Asterisk will support this. Obviously it won't work if your PRI provider hasn't enabled TBCT on your PRI. On the Asterisk side of things, you need to make sure you have facilityenable=yes and transfer=yes in zapata.conf for the bearer channels of your PRI. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Call deflection on PRI
Do you mean individual B-channels? That could be done in dialplan, with the ZapCD command... When its done that is :) Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: 27. februar 2008 16:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Call deflection on PRI Now to come up with a way to busy out individual channels via this or another method. This is one feature that is in great demand. Thanks, Steve Totaro On Wed, Feb 27, 2008 at 10:03 AM, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Actually this isnt the same as Two B-Channel transfer. This is done by sending a FACILITY message to the ISDN, which in hand then disconnects the call and sends it to the number provided in the call deflection message. All b-channels are closed when the FACILITY message is sent. You can also send a deflection reason code, such as Busy and Unavailable. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: 27. februar 2008 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Call deflection on PRI On Wed, 2008-02-27 at 08:50 -0500, Steve Totaro wrote: I recall that this is now part of Asterisk (1.4 or 1.6 or both). It really is a great feature rather than using two channels in trunk to trunk transfer. This is often called a Two B-Channel Transfer, or TBCT. As long as your PRI provider has this service enabled on your PRI, recent versions of libpri, zaptel, and Asterisk will support this. Obviously it won't work if your PRI provider hasn't enabled TBCT on your PRI. On the Asterisk side of things, you need to make sure you have facilityenable=yes and transfer=yes in zapata.conf for the bearer channels of your PRI. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 on PS3 Cell
Hello List. I just got my new PS3 yesterday, and first thing I did was of course to install Linux, and then compile asterisk, and it worked without any problems. My question is this... Is anybody looking into using the Cell processor for G729 enc/dec? Using the 6 SPE processing units available, you should be able to enc/dec a whole lot of channels at one time. Looking at the example code from IBM, for creating Cell-specific applications, it shouldn't be that big of a challenge to convert the G729 reference code, to Cell. If anybody is interested in trying this, any serious requests can get shell access to the machine, by contacting me off-list. Venlig Hilsen/Kind Regards Jon Leren Schøpzinsky ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf vs. AEL
Wouldnt that take a very large portion of datapower, to startup the parsers and such, instead of having the whole dialplan natively in Asterisk. We always try to do as much as possible in dialplan, so that we are not reliant on external scripts. Kind Regards Jon Leren Schøpzinsky -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: 3. oktober 2007 15:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] extensions.conf vs. AEL You have various scripting languages things like that can go in! /b On Oct 3, 2007, at 4:12 AM, Garth van Sittert wrote: Where would you suggest all the logic goes Brian? Garth Garth van Sittert BSc (Physics Computer Science) - Main: 08600 BITCO Phone: +27 (0)11 875 6900 Fax: +27 (0)11 875 6901 Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Web:www.bitco.co.za Brian West wrote: In my opinion the dialplan isn't where that logic belongs. /b On Oct 3, 2007, at 12:32 AM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and OCS integration
I would use SER or OpenSER as a middle man. Set it up to receive via TCP and send it on to the asterisk server using UDP. Kind Regards Jon Leren Schøpzinsky Solution Engineer Dansk Erhvervs-Telefon A/S tlf: +45 88200336 mob: +45 31206709 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dadsadsadf dsadasdsa Sent: 24. september 2007 13:29 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and OCS integration Hi List! does anyone played around with the OCS and Asterisk? I want to integrate OCS and Asterisk to enable Office Communicator 7.0 client to make and receive calls from PSTN I know that I need patch Asterisk to support TCP. But I am a bit ( a lot) lost Which more things should I need to keep in mind? Any advise will be wellcome :-) Thank you very much, Marta _ Acepta el reto MSN Premium: Protección para tus hijos en internet. Descárgalo y pruébalo 2 meses gratis. http://join.msn.com?XAPID=1697DI=1055HL=Footer_mailsenviados_proteccioninfantil ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR and MySQL
Another way to do this, is to use the func_odbc library. Its very good for production use, on larger sites. We use it on all of our asterisk servers, and it works great. Venlig Hilsen/Kind Regards Jon Leren Schøpzinsky Solution Engineer Dansk Erhvervs-Telefon A/S tlf: +45 88200336 mob: +45 31206709 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Sent: 14. august 2007 14:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR and MySQL On 8/14/07, Thiago Maluf [EMAIL PROTECTED] wrote: Hi Fabio, of course that you can. One way to do it is working with app MYSQL(), where you will put your sql as argumment. read more in http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL That's possible, but i wouldn't recommend on large production system. Using MySQL you would need to connect and disconnect all the time, and it takes resources.. I would suggest to append that info to CDR userfield (if you are storing your CDR in MySQL), and run periodically some script that extracts them. Of course it's more complex, but that would be my way. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astribank-8BRI
Hello We use the 2BRI version of Astribank in production, and it has been working non stop for about amonth now, without any problems. It was a bit difficult to setup, but other than that, it was great. Great concept with using the USB2 port for channel banks. Regards Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lars Bensmann Sent: 24. juli 2007 04:22 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Astribank-8BRI Hello, I'm in the process of building an Asterisk machine and need 5 or 6 BRI-Channels. I was looking for the beroNet and Junghans cards and stumbled upon the Xorcom Astribank xBRI products. Has anybody tried out the Astribank xBRI-Channel Banks? Are they production ready or should I go with a beroNet BN8S0 or JUNGHANNS.NET octoBRI ISDN? Thanks in advance, Lars -- We don't like their sound. Groups of guitars are on the way out. -- Decca Recording Company, turning down the Beatles, 1962 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.16/914 - Release Date: 23-07-2007 19:45 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.16/914 - Release Date: 23-07-2007 19:45 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with RNDIS
Hello List I am having some problems receiving RNDIS on a EuroISDN E1 in both Asterisk 1.2 and 1.4. Im not receiving anything, and when I do a pri debug span, I get this message: -- Making new call for cr 114 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 28 (cs0, Facility) Handle Q.932 ROSE Invoke component [ Handling operation 15 ] !! Unable to handle ROSE operation 15 [ 30 19 02 01 01 0A 01 02 A1 11 A0 0F A1 0D 0A 01 02 12 08 32 32 34 35 38 34 30 35 ] - [0..22458405] -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) q931.c:3294 q931_receive: call 114 on channel 28 enters state 6 (Call Present) q931.c:2570 q931_call_proceeding: call 114 on channel 28 enters state 9 (Incoming Call Proceeding) The 22458405 is the RDNIS that is supposed to be in the RDNIS field. Can anybody see why this is? Is it our operator that sends the information incorrectly? Kind Regards Jon Leren Schøpzinsky No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12-07-2007 16:08 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI Partial Re-Rounting
We are currently connecting to TeliaSonera in Denmark, and they said it should be supported via PRI supplementary services. I think their platform is Ericsson. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 6. juni 2007 22:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Partial Re-Rounting I think what they're talking about is forwarding the call before the call is established. If I remember correctly, it's call CF[U,B,NR] for call forward on unavailable, busy, and no response. Unfortunately though, none of the switchtypes support this variant of this function. However, if 2BCT is acceptable, we have a working implementation for DMS100 switchtype in 1.4. Matthew Fredrickson On Jun 6, 2007, at 9:53 AM, Eric ManxPower Wieling wrote: Jon Schøpzinsky wrote: Hello List We are trying to redirect calls directly, instead of opening a new channel and dialing out. Etc: A calls B on our asterisk, and is directly redirected to C We have been told that this feature should be available on a PRI level, and is called Partial re-routing. Anybody has an idea of whether this is supported in Asterisk? It is called 2BCT. It is supported on ATT 5ESS PRI lines. I don't think it is supported on NI2 or non-ATT switches. I've never used it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.472 / Virus Database: 269.8.11/837 - Release Date: 06-06-2007 14:03 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.472 / Virus Database: 269.8.11/837 - Release Date: 06-06-2007 14:03 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI Partial Re-Rounting
Hello List We are trying to redirect calls directly, instead of opening a new channel and dialing out. Etc: A calls B on our asterisk, and is directly redirected to C We have been told that this feature should be available on a PRI level, and is called Partial re-routing. Anybody has an idea of whether this is supported in Asterisk? Kind Regards Jon Schøpzinsky Detele. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.472 / Virus Database: 269.8.9/834 - Release Date: 05-06-2007 14:38 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What replaces SetCallerPres in 1.4
Hello SetCallerPres function seems to be removed from Asterisk 1.4. What function or application replaced it? Bit of a problem if you want to use CLIR on your PRI connections. Jon No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/815 - Release Date: 22-05-2007 15:49 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What replaces SetCallerPres in 1.4
Found the problem. I thought SetCallerXXX family of applications was retired in 1.4, so I didn't compile app_setcallerid. But seems that SetCallerId survived, and that SetCallerPres is located in the app_setcallerid.c Maybe somebody should move it to its own module. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: 23. maj 2007 15:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What replaces SetCallerPres in 1.4 Jon Schøpzinsky wrote: Hello SetCallerPres function seems to be removed from Asterisk 1.4. I have it in 1.4.4 drdos*CLI core show application setcallerpres drdos*CLI -= Info about application 'SetCallerPres' =- [Synopsis] Set CallerID Presentation [Description] SetCallerPres(presentation): Set Caller*ID presentation on a call. Valid presentations are: -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/815 - Release Date: 22-05-2007 15:49 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/815 - Release Date: 22-05-2007 15:49 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] MySQL/IVR Integration
Func_odbc is actually also backported to 1.2, so its your friend there too. Regards Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 21. maj 2007 08:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MySQL/IVR Integration in 1.4, func_odbc is your friend. Julian. David wrote: Hello, I'm looking to do the following, and I wonder if Asterisk can be used for it, and if yes, if anyone can point me to the relevant information (commands, sample config...): 1. Caller dials 111, 222 or 333. 2. Based on the dialed number, Asterisk queries an external MySQL table and retrieves alphanumeric data, plays/announces it to the user and deletes the row from the database: The SQL queries would look something like: SELECT user, pwd FROM codes WHERE dialed = '111'; DELETE FROM codes WHERE user=$user AND pwd=$pwd; Thanks, David Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email for dotr.com has been scanned by MessageLabs __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/813 - Release Date: 20-05-2007 07:54 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/813 - Release Date: 20-05-2007 07:54 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Conference server (or how to make a call withmore than 3 u
Use Snom phones. We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yehavi Bourvine +972-8-9489444 Sent: 19. marts 2007 09:14 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Conference server (or how to make a call withmore than 3 u On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote: Hello, On most SIP phones a conference call is done on the phone and is limited to 3 participants. Polycom phones has a configuration option to use a conference server instead of the internal conferencing feature. I guess I need some conference server; any experience with such a server which can interact with Asterisk? Why not use the MeetMe feature of asterisk? I need the person who initiated the conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Conference server (or how to make a call withmorethan 3 u
With 6 people it works, we have tried it. The 12 people is, as I said, only in theory, because, as you said, the CPU is probably not powerful enough. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: 19. marts 2007 09:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Conference server (or how to make a call withmorethan 3 u Jon Schøpzinsky wrote: Use Snom phones. We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. I don't think this is true. The Snoms do not have enough CPU power for 12 people in a conference *on the phone*. And I doubt that it works for 6. Does it? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Shoutcast music-on-hold
Hello List I am currently testing, using a shoutcast server as source for MOH. Here is the command im using: /usr/bin/wget -q -O - http://listen.coolfm.dk:80/ | /usr/local/bin/madplay -d -Q -z -o raw:- --mono -R 44100 - | sox -r 44100 -w -s -t raw - -r 8000 -c 1 -t raw - resample vol 0.10 I know that the normal examples, only shows using madplay without sox, but the quality is s bad when I do this, compared to using SoX to do the samplerate conversion. My problem is, that everytime somebody hangs up, and nobody is using the MOH, it seems as though it stops reading data from the shoutcast server. This results in the music re-buffering from the shoutcast server, which skips the music, and in this scenario results in a re-connect to the shoutcast server. Anybody know of a solution for this? Jon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Help - Poor Voice Quality
The part about 4569 being the IAX2 setup port, is not correct. All traffic, including RTP, travel through this port, when you use IAX. rtp.conf is used for SIP traffic, and possibly H232. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed W Sent: 8. februar 2007 11:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Help - Poor Voice Quality Hi Yes, I know that I am using IAX2 and not SIP for my connection to teliax. IAX2 is the preferred protocol for connection to teliax. I have the firewall configured to prioritorize port 4569 for IAX2. 1) 4569 is only the IAX setup port. Edit rtp.conf to limit the rtp ports to some subset and then prioritise those instead 2) Uplink bandwidth is always the constraint on these lines. This is highlighted in this case 3) Shorewall can't correctly prioritise bandwidth whenever using some kind of DSL service or whenever the packets are encapsulated such as the cable service. Read the linux QOS faq for more info and as a workaround slash the theoretical bandwidth in half in your shaping script. This should get you working and you can tweak later 4) Monitor the QOS buckets as you make/break calls to check that all the packets are classified correctly. Otherwise your voip packets might be accidently in the bulk box Basically VOIP goes from perfect to horrible when the jitter rises and packet loss goes up. Probably this is happening in your case Good luck Ed W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma card dying after 1hour
Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives packets from the ISDN, but when I make a call, I cant see it in wanpipemon, and asterisk just responds with a: NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) I am pretty shure that this is a configuration issue, but are there anything I need to be aware of when moving from a Digium card to a sangoma card? Kind Regards Jon Leren Schøpzinsky ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sangoma card dying after 1hour
I am running the newest version, from the sangoma wiki. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 10:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives packets from the ISDN, but when I make a call, I cant see it in wanpipemon, and asterisk just responds with a: NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) I am pretty shure that this is a configuration issue, but are there anything I need to be aware of when moving from a Digium card to a sangoma card? Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded as there are some resource leak fixes in that version. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sangoma card dying after 1hour
Asterisk is version 1.2.14, zaptel 1.2.12, libpri is whatever version was with zaptel 1.2.12 :) Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 12:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour Which asterisk versions etc etc? On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: I am running the newest version, from the sangoma wiki. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 10:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives packets from the ISDN, but when I make a call, I cant see it in wanpipemon, and asterisk just responds with a: NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) I am pretty shure that this is a configuration issue, but are there anything I need to be aware of when moving from a Digium card to a sangoma card? Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded as there are some resource leak fixes in that version. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with Digium TE410
Hello List Just want to check if anybody else is having this problem. Every time the PRI connections are disconnected, the card freezes, and I have to reload the driver, to make it work again. We are very seriously considering switching to Sangoma at this moment, due to this and other problems, but I want to know if there is a solution, and to make sure it isn't asterisk that's freezing the cards, and that the problem would re-appear on the sangoma cards. Kind Regards Jon Leren Schøpzinsky ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] AMAFlags alwaysDocumentation (or 3 in astcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill)
Hello Why not use the CDR(userfield) field instead. You can set that to any integer of your liking, and use that to identify the type of call. Jon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Keagy Sent: 8. januar 2007 06:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AMAFlags alwaysDocumentation (or 3 in astcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill) Is anyone out there using AMAFlags? I'd like to set this field as a marker to distinguish different types of calls in CDRs, but can't seem to make it respond to the documented commands Set(CDR(amaflags)=bill) or SetAMAFlags(bill). I've googled this issue, seen others have had this problem with IAX, with different DB drivers for CDR records, etc. I'm using SIP and LOCAL channels, asterisk-1.4beta2 release (I don't think upgrading to current release will fix this problem, it's been around for years based on trouble reports), both text .csv and mysql astcdr.cdr types. Seems like a problem with basic AMAflags support in CDR. They always show up as DOCUMENTATION in the .csv text file, and they always show up as '3' in mysql. I hurt my brain trying to follow the layers of indirection in the source code for where this is actually set. With verbosity turned on in asterisk console I can see the SetAMAFlags function being run. Any tips, tricks, or pointers in the right direction? Thanks, Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP peer lookup problems
Hello I am currently having a problem, that threatens to drive me insane... I cannot understand how Asterisk matches up a sip request with a peer. Here is my example: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.100.59:5060;branch=z9hG4bK00d97b99;rport From: 1088200336 sip:[EMAIL PROTECTED];tag=as54af3e4d To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: gw03 Max-Forwards: 70 Proxy-Authorization: Digest username=voipsip, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=, response=x, opaque= Date: Thu, 04 Jan 2007 17:38:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 1348 1349 IN IP4 192.168.100.59 s=session c=IN IP4 192.168.100.59 t=0 0 m=audio 11720 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (14 headers 11 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.100.59 : 5060 (NAT) Found peer '12345678' Jan 4 18:38:28 NOTICE[19589]: chan_sip.c:10469 handle_request_invite: Failed to authenticate user 1088200336 sip:[EMAIL PROTECTED];tag=as54af3e4d Jan 4 18:38:28 NOTICE[19589]: chan_sip.c:10469 handle_request_invite: Failed to authenticate user 1088200336 sip:[EMAIL PROTECTED];tag=as54af3e4d Why does Asterisk identify this request as coming from the user 12345678 and not from the user voipsip as its clearly stated in the Proxy-Auth string The user 12345678 is registered on the server, coming from the same IP as voipsip, but if voipsip is a different user, why on earth does asterisk not identify it as voipsip instead of 12345678??? Some of the values and numbers are changed for security, etc. nonce=, response=x and such. Kind Regards Jon Leren Schøpzinsky ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fonebridge2
Hello List Does anybody have any experience with the FoneBridge line of products from RedFone? I think their HA implementation sounds interesting, and like the prospect of having dedicated hardware for our PRI connections. Kind Regards Jon Leren Schøpzinsky ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WRAP+astlinux g729
Hello How many simultaneous conversations g.729a should one expect with a WRAP board running Asterisk? Has anybody tried this? Kind Regards Jon Leren Schøpzinsky ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Standardized IVR UI Pattern (was: Re: [asterisk-users] Is thereany Asterisk controllable thermostat?)
If you want a standardized ivr ui pattern, wouldn't something like VoiceXML be interesting? That's a standard for use with IVR applications. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: 7. december 2006 15:53 To: Asterisk-Users Subject: Standardized IVR UI Pattern (was: Re: [asterisk-users] Is thereany Asterisk controllable thermostat?) On Wed, 2006-12-06 at 23:51 -0700, [EMAIL PROTECTED] wrote: Date: Wed, 06 Dec 2006 22:37:01 -0500 From: Steve Prior [EMAIL PROTECTED] Subject: Re: [asterisk-users] Is there any Asterisk controllable thermostat? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Doug Crompton wrote: and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. Doug I've started to play with writing some code using the Java FastAGI interface to connect to my home automation system. The code is working and I could now write whatever I wanted, but I haven't figured out what would be a reasonable menu interface that wouldn't be very annoying to use. I'd be very interested to hear what menu structures and what actual capabilities people have found useful and nice to use. For example, has anyone come up with something less annoying than the following dialog: Press 1 for living room, press 2 for outside, press 3 for bedroom (I press 2) Press 1 for porch light, press 2 for garage light (I press 1) Press 1 to turn on, Press 2 to turn off, Press 3 to say current status (I press 1) congratulations, you just spent several minutes just to turn on a light! I don't know why IVR menus still include so much extra verbiage. They should act like numbered lists - everyone knows the stated number means the key to press, and the stated name means what you will get. So: (Listens for DTMF) Hello, this is home thermostat. 1 living room 2 outside 3 bedroom (waits for DTMF, maybe repeats after a 2 second pause) (I press 2) (Listens for DTMF) Outside 1 porch light 2 garage light (waits for DTMF, maybe repeats after a 2 second pause, offers to hangup after maybe 15 seconds) (I press 1) (Listens for DTMF) Outside Porch light 1 on 2 off 3 say current status (waits for DTMF, maybe repeats after a 2 second pause) (I press 1) (Listens for DTMF) Outside porch light status turned on star for options, hash to hangup (waits for DTMF, maybe repeats after a 2 second pause) That menu system would take about 10 seconds the first time through, listening to all prompts. Subsequent navigation could take 2-4 seconds. Subsequent shortcuts through a collapsed star-hash menu could take 1-2 seconds. Make the star key an escape key to the previous scope. Make the hash key an Enter key that terminates any multiple-key entry. Collapse all menu scopes/items into a single long list that can be reached at any time through star-hash. Introduce the whole menu system with press star for options, to the star-star menu. Make the 0 option in the star options menu the path to a human operator, if there is one. And always immediately feedback to any received key with at least a click. This simple UI should be common to every IVR app, so anyone can always use it without listening for a while to learn how to navigate the IVR. In fact, I call this system IKR (Interactive Key Response), and maybe every system should answer the call with first saying IKR. Then callers would immediately know when our skills on the common UI would work, without waiting to learn, or mistake it. If the server played a few touchtones, like 4-5-7 (keypad IKR) while saying IKR, smart automated clients could detect the system and use it. To complete the interactivity protocol, every spoken digit to be pressed in the numbered menus would also play the digits' DTMF. And the intro to the scope to which a client DTMF navigated would play the last digits that navigated there from the previous scope while saying the name of the new scope. This is the system that I used to use when I built dedicated IVR systems a dozen years ago (on Dialogic HW). Almost no IVR people were on the Internet then, before the Web. There was no community, and IVR vendors competed so harshly that they couldn't get such a standard interface going, even for mutual benefit. So now everyone hates using IVR, even when it's better than a human operator. And we still all roll our own from scratch. But with Asterisk, and web/maillists connecting a community, we can adopt a common system. If enough people like it, I will publish the spec, and maybe write the RFC. Or maybe there's a better one that will be adopted more widely more quickly, and we can get behind that. If you don't like it, you can
RE: [asterisk-users] 200+ analog phones connected to FXS modules
I think It would be cheaper to use dedicated VoIP PSTN Gateways, such as audiocodes or similar (audiocodes is actually a bad example, as their not that cheap). But dedicated ATA hardware with 24 or more ports. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vieri Sent: 30. november 2006 10:15 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 200+ analog phones connected to FXS modules I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 200+ analog phones connected to FXS modules
I would just guess that the PCI bus would be pretty busy, with 3 T1 cards. Couldn't that be a problem? Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 30. november 2006 12:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] 200+ analog phones connected to FXS modules You could put at least two Rhino quad t1 cards and that would give you 8 times 24 ports and I heard of one with those cards plus a dual t1 card which is 240 extensions on one server. this would take up 3 pci slots. on Thursday 11/30/2006 Jon Schøpzinsky([EMAIL PROTECTED]) wrote I think It would be cheaper to use dedicated VoIP PSTN Gateways, such as audiocodes or similar (audiocodes is actually a bad example, as their not that cheap). But dedicated ATA hardware with 24 or more ports. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vieri Sent: 30. november 2006 10:15 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 200+ analog phones connected to FXS modules I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] Dropping Connections
Helo My money is on the WLAN part of the equation. We actually dropped WLAN SIP phones altogether, since they worked so poorly. Connection loss, bad audio quality and low coverage range. Just my 5 cents... Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Mike Heininger Sendt: 10. november 2006 12:57 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] Dropping Connections Hi! We have an installation with WLAN SIP phones only. Sometimes we have connection drops. What is the best way to debug if we have problems with the WLAN or the SIP devices or the uplink to the IAX Provider. TIA, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.14.1/527 - Release Date: 09-11-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.14.1/527 - Release Date: 09-11-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] ip address in CDR
You can use the CDR(userfield) value, to save the ip's in the CDR record. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Benjamin Jacob Sendt: 3. november 2006 06:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] ip address in CDR Hello ppl, Any way to store the origination or termination IP addresses in CDRs? cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.13.23/513 - Release Date: 02-11-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.13.23/513 - Release Date: 02-11-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choice of soundfile format
Hello What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? Kind Regards Jon Leren Schøpzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 - Release Date: 24-10-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] Screen pop based on incoming DID
Hello I would think that using the manager interface, would be the easiest way of implementation. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Greg Delgado Sendt: 3. oktober 2006 14:44 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] Screen pop based on incoming DID I want to pop up a web page when a queue member phone rings but, instead of displaying the clid, I want to display the DID number the call came in. Any ideas how to best implement this? Greg __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.407 / Virus Database: 268.12.12/461 - Release Date: 02-10-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.407 / Virus Database: 268.12.12/461 - Release Date: 02-10-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] txfax reliability on TDM cards
Hello Use IAXmodem+hylafax instead. It works a lot more stable than rxfax and txfax. Probably something to do with hylafax being more accepting of errors. Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Jerry Geis Sendt: 28. september 2006 15:51 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] txfax reliability on TDM cards Hi all, What is the reliability of sending faxes with txfax? I am sending a 4 page fax. I have received 1 and 2 pages but never the whole thing? Do I have to have T1 or something different to reliably send faxes with a TDM card? Jerry -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.12.9/457 - Release Date: 26-09-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.12.9/457 - Release Date: 26-09-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with app_directed_pickup
Hello List I am having a strange problem, that seems to have appeared from nothing. Im running Asterisk 1.2.9.1, and we use the app_directed_pickup application. But recently we get this result: AGI Script Executing Application: (Pickup) Options: (882003308820033188200332882003338820033488200335882003368820033788200341) Sep 8 11:34:19 NOTICE[19819]: app_directed_pickup.c:125 pickup_exec: No call pickup possible for 88200330... Sep 8 11:34:19 NOTICE[19819]: app_directed_pickup.c:125 pickup_exec: No call pickup possible for 88200330... Sep 8 11:34:19 NOTICE[19819]: app_directed_pickup.c:125 pickup_exec: No call pickup possible for 88200331... Sep 8 11:34:19 NOTICE[19819]: app_directed_pickup.c:125 pickup_exec: No call pickup possible for 88200331... Sep 8 11:34:19 NOTICE[19819]: app_directed_pickup.c:125 pickup_exec: No call pickup possible for 88200332... Sep 8 11:34:19 NOTICE[19819]: app_directed_pickup.c:125 pickup_exec: No call pickup possible for 88200332... Sep 8 11:34:19 NOTICE[19819]: app_directed_pickup.c:125 pickup_exec: No call pickup possible for 88200333... Sep 8 11:34:19 NOTICE[19819]: app_directed_pickup.c:125 pickup_exec: No call pickup possible for 88200333... It seems as though the app_directed_pickup application is not iterating properly trough the list of numbers. Anybody has an idea for a fix? Kind regards Jon Leren Schøpzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.12.2/441 - Release Date: 07-09-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] E61
I also have this phone, and have stumbled in to the same problem. I just think that it isn't in nokia's interest to change this, as it forces consumers to have some sort of local hardware, that (possibly) only the telecom provider can give them. This forces the users away from using cheaper services. Nokia makes a load from the telecom operators around the world, and are not interested in pissing them off, by letting their users bypass their price structure. Just my 5 cents. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Andreas Sikkema Sendt: 24. august 2006 15:24 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [asterisk-users] E61 Anyone here use the Nokia E61 ? I am looking to invest in a wifi phone and I want to get the best. Is it good as far as reception ? That is of most importance to me. Thanks. I've tried it in the last couple of days. The biggest issue for me ist that it HAS to be on the same side of a NAT as the server it talks to (asterisk, ser, etc). If it is on the private side of a NAT and the server is on the public side, it doesn't work. I've read something on the Nokia forums that Nokia is aware of the problem and it will be solved. My problem is that they want to solve this using STUN etc, while I would prefer they also wouldn't have the software care if it is on the inside of a NAT like most other CPE's so our platform can take care of things. -- Andreas SikkemaBBeyond Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.5/426 - Release Date: 23-08-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.5/426 - Release Date: 23-08-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] Hint extension issue - bug?
Hello Wouldn't the correct way of handling call limits, be using the Call Group Applications available in Asterisk? Regards Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Andrew Kohlsmith Sendt: 23. august 2006 15:30 Til: asterisk-users@lists.digium.com Emne: Re: [asterisk-users] Hint extension issue - bug? On Wednesday 23 August 2006 08:48, Watkins, Bradley wrote: It's not a bug. When you use type=friend, it will create a user object *and* a peer object. This will make call-limit not function, thereby breaking hints. There is no reason to use friend anyway. It does not gain you any functionality, and in fact breaks some. This is broken behaviour. I don't know why we have the distinction of users and peers in the first place. A single entry with something along the line of calltype taking incoming, outgoing or both would be far clearer and eliminate all this inconsistency. Call-limiting not working with users is just as dumb an idea as users not being able to trunk calls in iax2. They're artificial boundaries set up for no reason other than to force a distinction between the two types of entries. Eliminate all the crap and let me use the damn PBX how I want; that's one of the biggest features of Asterisk. Stop trying to protect me for my own good. Document the shit, make it consistent and let the community support the clueless. You don't see this kind of crap with apache, openswan, postfix or even the kernel itself. There's no need to tie my hands behind my back in order to protect the newb. All you'll end up with is a system only newbs want to use. Before anyone accuses me of not putting my money where my mouth is: I've submitted a number of patches over the years to correct or address what I consider inconsistencies, and I do what I can to test out trunk, report bugs and document. I'm doing what I can to help the system. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.5/425 - Release Date: 22-08-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.5/425 - Release Date: 22-08-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] lbProxy
Hello Ive been trying to use the lbProxy SIP load balancing proxy. After I actually got it compiled, using the CMSOFAZ.COM version, I began experimenting. I quickly ran into a problem. Heres my setup: wan--|lan Phone - lbProxy - Asterisk lbProxy sends all of the sip packets to Asterisk, but when asterisk responds, it chooses the port from which the request came, instead of using the port number in the Via statement. This is a good thing normally, due to NAT, but even if I set nat=no or nat=never, it still responds to this port instead of the port in the Via Statement. Has anybody gotten this to work, and can explain how. Im about 1 day away from writing my own load balancing sip proxy, but would love if I could use lbProxy instead :) Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.4/424 - Release Date: 21-08-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] IAX trunk behing NAT with dynamic IP
Set the host=dynamic on serverA, and let the serverB register with serverA Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Andre Courchesne - Consultant Sendt: 8. august 2006 14:09 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] IAX trunk behing NAT with dynamic IP Hi, Ok, I got a working setup where the * server having the telephony card has a fixed internet IP address (serverA). I am using an IAX trunk from this server to an other one which has a dynamic IP address and is behind a NAT firewall (serverB). Everything works fine untill serverB internet IP address changes. My host line is set to a dynamic DNS entry (with zoneedit.com) How can I resolve this so that serverA see the IP address change of serverB ? Andre Courchesne - Consultant http://www.net-forces.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.7/411 - Release Date: 07-08-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.7/411 - Release Date: 07-08-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 Trunking CPU usage
Hello Im trying to decide whether or not I want to use IAX2 trunking on our WRAP based customer computers. As it only has a 200mhz processor, I want to make shure that the trunking part does not affect call quality. Does anybody know if trunking is more CPU intensive than non trunking? Regards Jon Leren Schøpzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.5/406 - Release Date: 02-08-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] VOIP phone for Receptionist use
For the VoIP phone question, I can warmly recommend the Snom 360. When using hints in asterisk, this is the perfect phone for secretary use, as you can also add a side panel with 48 extra buttons with lights. When using hints you can see when extensions a talking, ringing, as well as have up to 12 ingoing lines. A very good phone, that we recommend to all of our large customers, with a secretary. Regards Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Jeff Busch Sendt: 2. august 2006 02:20 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] VOIP phone for Receptionist use I've searched through the newsgroup and online and haven't found an answer for my question... maybe I am looking for the wrong terms, I am not sure... I have a client that would like a phone that is like a typical receptionists phone. Requirements: - Ability for their3 lines to light-up a button on the phone when one of them rings in. - Ability for the phone to ring when the receptionist is on one call and a second or third call is incoming. (this has been the biggest frustration up to now. When a second call comes, there is no tone that heard on the IP500. Perhaps I am missing a setting?) We are currently using: Asterisk @ Home 2.1 Polycom IP500/501 phones Is there a way to do what we need to using the IP500 phones? If so, can anyone give me instructions on how to make it work with [EMAIL PROTECTED]? Thanks for your help in advance. Jeff -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.5/404 - Release Date: 31-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.5/404 - Release Date: 31-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] Sip phone settings set when user registers
Hello Just use Snom or grandstream phones. They can be provisioned very easily via HTTP. You just setup a config URL on the phones, and they get their configurations from there. If you want to get more advanced, they can send along their MAC address, and thereby enabling you to custom config them directly from a central application, based on the phones MAC address. The snom phones can even be instructed to download a configuration from a URL via DHCP. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Nik Engel Sendt: 27. juli 2006 10:39 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] Sip phone settings set when user registers Hi all ! I am planing to set up around 20 SIP Phones which will be purchased in one bunch, I am more or less free of choice. I wonder if anyone knows sip phones which allow configuration upon login. The following scenario: User logs into any phone and the settings of the phone are always the same. Meaning individual key assignement is always the same. Is this possible with asterisk in combination which any phone or do I require special phones. Thanks for any advices Nik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/401 - Release Date: 26-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/401 - Release Date: 26-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 trunking problems
Hello list We are having some strange problems. When we setup trunking between two of our servers, the connection only uses trunking one way. Ex: Data From callingserver to receivingserver uses trunking Data from receivingserver to callingserver does not use trunking. I discovered this problem by looking at a tcpdump in Ethereal, and I can see that the trunked meta packets only goes one way. The other way uses normal Mini packets with raw a-law data. Heres the configurations, with password, username and server info removed. Callingserver: [gsmgw1] secret=*** username=** host=*** type=peer trunk=yes notransfer=yes disallow=all allow=alaw allow=g726 Receivingserver: [**] secret=*** context=default host=** type=user accountcode= trunk=yes notransfer=yes Both servers have ztdummy module installed and loaded. Regards Jon Schøpzinsky -- Internal Virus Database is out-of-date. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.0/388 - Release Date: 13-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 vs TDMoE
Hello List We are having some load problems, and they are impacting IAX2 performance the most, with large amounts of jitter and lost packets. I'm currently thinking about using TDMoE for internal communication between our Asterisk servers. Does anybody know how load problems impact TDMoE? We are not having quality problems on our E1 connections, so I would guess that performance should be the same for TDMoE. Kind Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.10/387 - Release Date: 12-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 trunking problems
Hello list We are having some strange problems. When we setup trunking between two of our servers, the connection only uses trunking one way. Ex: Data From callingserver to receivingserver uses trunking Data from receivingserver to callingserver does not use trunking. I discovered this problem by looking at a tcpdump in Ethereal, and I can see that the trunked meta packets only goes one way. The other way uses normal Mini packets with raw a-law data. Heres the configurations, with password, username and server info removed. Callingserver: [gsmgw1] secret=*** username=** host=*** type=peer trunk=yes notransfer=yes disallow=all allow=alaw allow=g726 Receivingserver: [**] secret=*** context=default host=** type=user accountcode= trunk=yes notransfer=yes Both servers have ztdummy module installed and loaded. Regards Jon Schøpzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.10/385 - Release Date: 11-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] setting up an email to fax with asterisk
Hello If you look at hylafax.org, you can find several windows clients for Hylafax. http://www.hylafax.org/content/Desktop_Client_Software Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Erick Perez Sendt: 10. juli 2006 09:52 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] setting up an email to fax with asterisk So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo) and then how the windows clients send email-to-fax to the above machine? On 7/10/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. You need to install an iaxmodem on the machine where the hylafax server is installed. Which can probably be the Asterisk server. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] How to collect Call duration, Dialout Call files?
Hello Just use NoCDR() in the non bridged local context. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Marco Mouta Sendt: 7. juli 2006 11:22 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] How to collect Call duration, Dialout Call files? Hi all, I've been planing to implement a webcall portal to dial SIP extensions from my pbx, I've implemented this with dialout call files. Could you advice me on the best way to collect call duration of this calls, only this way i can allow my users to place external outgoing calls. I've need to use local_chan to avoid CDR missing details. But this way i get CDR of two calls, and what I wanna get is the call duration of the bridged call. By default i get 2 calls in CDR, instead of the bridged final call. Is it much better to use Asterisk Manager API instead of Dialout Call files? Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] How to collect Call duration, Dialout Call files?
When I used the .call files, I made so that the call went through a Local extension, where I didn't record the call, so that it would only be logged on the outgoing channel Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Marco Mouta Sendt: 7. juli 2006 13:55 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] How to collect Call duration, Dialout Call files? Sorry i didn't get your idea. could you explain me what you mean? Are you saying to make CDR in only one of the legs? Best regards, Marco Mouta On 7/7/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello Just use NoCDR() in the non bridged local context. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Marco Mouta Sendt: 7. juli 2006 11:22 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] How to collect Call duration, Dialout Call files? Hi all, I've been planing to implement a webcall portal to dial SIP extensions from my pbx, I've implemented this with dialout call files. Could you advice me on the best way to collect call duration of this calls, only this way i can allow my users to place external outgoing calls. I've need to use local_chan to avoid CDR missing details. But this way i get CDR of two calls, and what I wanna get is the call duration of the bridged call. By default i get 2 calls in CDR, instead of the bridged final call. Is it much better to use Asterisk Manager API instead of Dialout Call files? Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] How to collect Call duration, Dialout Call files?
I switched to using the Manager API, as I thought it was easier to use. I use it from a PHP script, which uses the flaAPI.php class. Just use the originate action, set the channel to a Local channel and connect to a context. Are both calls going out over PSTN, or are one of them going via SIP? -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Marco Mouta Sendt: 7. juli 2006 14:15 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] How to collect Call duration, Dialout Call files? did u try asterisk manager api? On 7/7/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: When I used the .call files, I made so that the call went through a Local extension, where I didn't record the call, so that it would only be logged on the outgoing channel Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Marco Mouta Sendt: 7. juli 2006 13:55 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] How to collect Call duration, Dialout Call files? Sorry i didn't get your idea. could you explain me what you mean? Are you saying to make CDR in only one of the legs? Best regards, Marco Mouta On 7/7/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello Just use NoCDR() in the non bridged local context. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Marco Mouta Sendt: 7. juli 2006 11:22 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] How to collect Call duration, Dialout Call files? Hi all, I've been planing to implement a webcall portal to dial SIP extensions from my pbx, I've implemented this with dialout call files. Could you advice me on the best way to collect call duration of this calls, only this way i can allow my users to place external outgoing calls. I've need to use local_chan to avoid CDR missing details. But this way i get CDR of two calls, and what I wanna get is the call duration of the bridged call. By default i get 2 calls in CDR, instead of the bridged final call. Is it much better to use Asterisk Manager API instead of Dialout Call files? Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] B2BUA Webbased and Click 2 dial apps
Hello You can just use the Asterisk Manager API. Its relatively easy to create this kind of application, just look at the Originate function of the API. http://www.voip-info.org/wiki/view/Asterisk+manager+API Theres lots of examples for many different programming languages. Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Dinesh Sendt: 6. juli 2006 10:41 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] B2BUA Webbased and Click 2 dial apps Hello, I have a requirement of bridging 2 sip connections via asterisk, which has to be web based. A person has to go to a webpage and enter his from sip uri(say sip1) and enter another sip uri(say sip2). Upon pressing the connect button, the webpage needs to send say a dial sip1 uri and dial dip uri 2 and bridge the call? Do I need any special sip api for this? Any ideas will be niceJ. Does this webpage has to be on asterisk server running on the machine? Or can it be passed as a string to the server from the webserver? Regards, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : [EMAIL PROTECTED] WWW: www.imcb.a-star.edu.sg -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.9/382 - Release Date: 04-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Nokia E61
Hello This isn't possible as the phone does not support NAT. You have to have a local Asterisk, or a SIP proxy on your local network. This Nokia feature is probably to prevent normal users using IP telephony, because it would hit the normal mobile providers. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Amund Nygaard Sendt: 5. juli 2006 09:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: SV: [Asterisk-Users] Nokia E61 Hello Has anyone tried a Nokia E6x phone when it is natted? Like behind a hotspot or similar? BR Amund Nygaard -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Devraj Mukherjee Sendt: 4. juli 2006 12:49 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Nokia E61 Thanks guys. How about the quality of the call etc? Are you happy with the phone, do you recommend them? On 7/4/06, Antonio Rabena [EMAIL PROTECTED] wrote: Hi, configuration for E61 is the same as E60. As for the codec, G729 works between E60/61 phones (G729 passthru). At 03:44 PM 7/4/2006, you wrote: Devraj Mukherjee wrote: Hello world, Any success stories of getting a Nokia E61 to work with Asterisk server? Interested to hear before we buy them for work :) I don't know about e61, but I connected an e60 up yesterday that wasn't any hassle. Even the stories about poor quality with WPA + G.729 seemed to be false. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.8/381 - Release Date: 03-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.8/381 - Release Date: 03-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Error in config sample for GoToIf?
Hello
As far as ive understood, you can just write
Exten = s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail)
${AVAILSTATUS} would return 1, and ${AVAILSTATUS} would return 1
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Brian Capouch
Sendt: 27. juni 2006 09:10
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Error in config sample for GoToIf?
My teeth are on edge after this one. A couple of perfectly good hours
of my life, and I still don't know what's going on. . . .
The extensions.conf.sample that comes with the current SVN trunk has
this line, in an example that shows how to use ChanIsAvail:
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
I couldn't get this to work unless I surrounded the first part of the
test with quotes, too, like this:
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
Leaving aside the completely separate madness of trying to determine
just what values mean what for the variable $AVAILSTATUS (which I would
be glad to receive a pointer to), is it indeed the case that the example
in the distribution is in error, or is there some other subtle rule that
is causing the behavior of this line to be correct with the extra quotes
but incorrect otherwise?
Thanks.
B.
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SV: [Asterisk-Users] periodic-announce not working
Hello
You have announce-frequency = 0
That would mean no announcements.
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Michiel van Baak
Sendt: 22. juni 2006 16:08
Til: asterisk-users@lists.digium.com
Emne: [Asterisk-Users] periodic-announce not working
Hi all,
I use asterisk 1.2.6 with queues.
This is my queue entry in queues.conf:
[460]
strategy = ringall
servicelevel = 60
context = reception
timeout = 25
retry = 2
maxlen = 0
announce-frequency = 0
periodic-announce-frequency = 25
announce-holdtime = no
periodic-announce = bovendonk/phone-queue
joinempty = yes
leavewhenempty = no
member = SIP/460
No matter how long I stay in the queue, it is never playing
the file to me.
This is the dialplan that puts me in the queue:
exten = _4XX-wachtrij,1,Verbose(1,phone busy so go to
queue)
exten = _4XX-wachtrij,n,Answer()
exten = _4XX-wachtrij,n,Playback(bovendonk/phone-queue)
exten = _4XX-wachtrij,n,Queue(${EXTEN:0:3}|tr||500)
Please help
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GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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[Asterisk-Users] IAX2 Dial command
Hello
I am trying to use this command to dial an IAX2 channel, with a supplied
context, etc:
Dial(IAX2/myiax2peer/[EMAIL PROTECTED])
This fails, with an authentication failed message while:
Dial(IAX2/myiax2peer/${EXTEN}) succeeds with out a hitch.
Why is this???
Regards
Jon
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[Asterisk-Users] SIP Registrations and DUNDi
Hello list I've been implementing an asterisk based cluster, and are having grave problems with SIP. My current implementation monitors registrations through the Asterisk Manager interface, but it seems to not register all registrations. Because of this, ive been looking at DUNDi, to implement the cluster. Has anybody done this with success? It seems that it uses the extensions instead of the SIP accounts, will this then work with Realtime? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.0/366 - Release Date: 15-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF when using g.729
Hello How do I get Asterisk to receive DTMF from our Snom phones, when I use G.729? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] DTMF when using g.729
I should note that we are not running the Digium g729 implementation, but the intel one. Also, to not angry people, this ofcourse isn't used in our production environment, only for testing if we want g.729. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Moises Silva Sendt: 14. juni 2006 15:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] DTMF when using g.729 Is new to me that using G729 codec is a problem when sending DTMF. Could it be that you are a little bit confused? Usually the problems with DTMF depend on how the phone is configured and how Asterisk is configured (DTMF using SIP INFO, RFC2833 etc), check this out: http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode Regards. On 6/14/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello How do I get Asterisk to receive DTMF from our Snom phones, when I use G.729? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Vs SIP cpu load
Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call status subscriptions on multiple servers
Hello List Is there a way to have hints sent between multiple servers? We are currently implementing a cluster solution for our asterisk servers, and the problem is this. User A registers on Asterisk 1 and user B registers on Asterisk 2. User A subscribes to user B's status, through SIP NOTIFY messages. As user B is registered to Asterisk 2, and not Asterisk 1, the NOTIFY messages are only sent on Asterisk 2, and Asterisk 1 does not know the status of user B. Is there a way to replicate subscription info between asterisk servers? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Database file to copy for active sessions.
Hello I can save you a lot of time, and tell you that it wont work. It does hold some registration information in the asterisk database, but most of the information is kept internally in Asterisk. Just FYI. Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Shenen Shenen Sendt: 9. juni 2006 11:37 Til: asterisk-users@lists.digium.com Emne: [Asterisk-Users] Database file to copy for active sessions. How can I copy all the contenent of the asterisk database to another machine? I want copy all the active sessions from one [EMAIL PROTECTED] to another one and running on the second(thisI can do using vrrp protocol, it isn't a problem), I want copy onlyall the active sessions and softphone registrations to another [EMAIL PROTECTED] and then run on it. -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Database file to copy for active sessions.
Its a little more tricky than that. Our solution involves an external manager application, some clever IAX2 routing and dialplan mysql queries. We tried the solution with just copying the registration, but it seems as though the SIP channel has the registry information in an Internal data structure. Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Shenen Shenen Sendt: 9. juni 2006 11:56 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Database file to copy for active sessions. ok...but if I run a softphone and it is registered in the CLI and I see this: -- Registered SIP '655' at 192.168.251.10 port 1175 expires 900 this registration where is put?in which file? Can I copy this registration to another machine? On 6/9/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello I can save you a lot of time, and tell you that it wont work. It does hold some registration information in the asterisk database, but most of the information is kept internally in Asterisk. Just FYI. Jon Fra: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] På vegne af Shenen Shenen Sendt: 9. juni 2006 11:37 Til: asterisk-users@lists.digium.com Emne: [Asterisk-Users] Database file to copy for active sessions. How can I copy all the contenent of the asterisk database to another machine? I want copy all the active sessions from one [EMAIL PROTECTED] to another one and running on the second(thisI can do using vrrp protocol, it isn't a problem), I want copy onlyall the active sessions and softphone registrations to another [EMAIL PROTECTED] and then run on it. -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Database file to copy for active sessions.
There is a solution, but its not straight forward, and not really documented anywhere. A possible solution, is to set a SER server up, before your asterisk, and let that handle the SIP registrations. Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Shenen Shenen Sendt: 9. juni 2006 12:21 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Database file to copy for active sessions. Somy only solution is to use only X-lite softphone where I can add more than 1 proxy, and a Cisco switchboard where I can set up a VRRP protocol, so in case of fall, the cisco make the resolutions of all tables and permited me to call from IP phones like CISCO IP phones or wi_fi phone without problems or registration in asterisk.I think..becouse in this way I see there isn't a solutionright? On 6/9/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: It's a little more tricky than that. Our solution involves an external manager application, some clever IAX2 routing and dialplan mysql queries. We tried the solution with just copying the registration, but it seems as though the SIP channel has the registry information in an Internal data structure. Jon Fra: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] På vegne af Shenen Shenen Sendt: 9. juni 2006 11:56 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Database file to copy for active sessions. ok...but if I run a softphone and it is registered in the CLI and I see this: -- Registered SIP '655' at 192.168.251.10 port 1175 expires 900 this registration where is put?in which file? Can I copy this registration to another machine? On 6/9/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello I can save you a lot of time, and tell you that it wont work. It does hold some registration information in the asterisk database, but most of the information is kept internally in Asterisk. Just FYI. Jon Fra: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] På vegne af Shenen Shenen Sendt: 9. juni 2006 11:37 Til: asterisk-users@lists.digium.com Emne: [Asterisk-Users] Database file to copy for active sessions. How can I copy all the contenent of the asterisk database to another machine? I want copy all the active sessions from one [EMAIL PROTECTED] to another one and running on the second(thisI can do using vrrp protocol, it isn't a problem), I want copy onlyall the active sessions and softphone registrations to another [EMAIL PROTECTED] and then run on it. -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] TSP on linux
Yes There is AstTapi: http://www.voip-info.org/wiki/view/AstTapi Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af [EMAIL PROTECTED] Sendt: 9. juni 2006 12:32 Til: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Emne: [Asterisk-Users] TSP on linux Hi, Can anybody tell me, is their a tsp for asterisk on linux ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Call status subscriptions on multiple servers
Can you then inform me on what structures this information is stored in, in the asterisk code? Then ill try to do a quick dirty version of the replication. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Kevin P. Fleming Sendt: 9. juni 2006 16:25 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Call status subscriptions on multiple servers - Jon Schøpzinsky [EMAIL PROTECTED] wrote: Is there a way to replicate subscription info between asterisk servers? Not at this time, no. That will be probably be worked on during the next development cycle. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: SV: [Asterisk-Users] I can hear only one way when I use nokiae-60withX-lite
Thats just the thing, and it sucks, because the VoIP implementation actually works very good. Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af list mail Sendt: 8. juni 2006 02:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] I can hear only one way when I use nokiae-60withX-lite Sounds like they crippled the phone for cellulars sake. On Jun 7, 2006, at 10:35 AM, Jon Schøpzinsky wrote: Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work. That means, that you cant just walk down to your local airport, and use the IP part of the phone on their network. You have to have a non nat local server, to get it to run. Other than that, the phone can accept calls both from cellular network and IP network, and actuatly works quite well, both for cellular and IP traffic. But you cant do seamless handover, for example when you walk out of the building. You have two different numbers, your mobile number and your IP number And these cant automaticly be transferred. Hope this answeres your question Regards Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] På vegne af Olivier Krief Sendt: 7. juni 2006 16:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite 2006/6/7, Jon Schøpzinsky [EMAIL PROTECTED]: Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon What do you mean by users has to have some local equipment from the telco ? Do you think Nokia E60, E61 and E70 are appropriate for Fixed Mobile Convergence (each mobile phone being reachable at the same time from inhouse PBX and Telco's mobile network without any handover or roaming between both networks) ? Regards -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using regcontext
Hello List Ive been trying to use regcontext, but I cant get it to work. Ive setup my sip peers to have the regexten _[0-9]., so that I can capture all registrations in a single extension. But when they register, I can see that the dynamic extension is created, but none of the rest of the code is executed, priority 2-4. Can anyone explain how I should use the regcontext parameter, etc. am I using it wrongly? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/358 - Release Date: 07-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Using regcontext
Hello Thanks for the answer... Just realized it myself, as your mail arrived :) Could be a nice feature though. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Olle E Johansson Sendt: 8. juni 2006 12:09 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Using regcontext 8 jun 2006 kl. 11.57 skrev Jon Schøpzinsky: Hello List Ive been trying to use regcontext, but I cant get it to work. Ive setup my sip peers to have the regexten _[0-9]., so that I can capture all registrations in a single extension. But when they register, I can see that the dynamic extension is created, but none of the rest of the code is executed, priority 2-4. Can anyone explain how I should use the regcontext parameter, etc. am I using it wrongly? You can't set aq regexten= setting to a wildcard. Regexten does not capture registrations, it adds an execution step to an exact extension. Regards, /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/358 - Release Date: 07-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/358 - Release Date: 07-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 channel problems
Hello List We are a VoIP telco, running Asterisk. We have been having problems with our IAX2 channels for some time now. Our problems are jitter, and lost packets, resulting in bad audio quality. The weird thing is, that this mostly occurs on our local network. We have tested the network with pinging an hour, without any lost packets. One of our customers also has problems using IAX2, and he is only two networks away, according to traceroute. He is on a 100mbit dedicated connection. Is there a general problem in the IAX2 channel, which causes jitter? We are running 1.2.9.1, and have tried 1.2.7.1, 1.2.5 and 1.2.0, and we have the same problems with all of them. Our average system load is around 2-3, and we have 905 registered sip users, and around 60 calls running at all time, to queues, SIP and Zap channels. Regards Jon Schoepzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] I can hear only one way when I use nokia e-60 withX-lite
Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af John Joseph Sendt: 7. juni 2006 13:59 Til: Asterisk Users Emne: [Asterisk-Users] I can hear only one way when I use nokia e-60 withX-lite Hi I am facing some problems in making calls to Nokia E60 ,from other sip extensions, I am able to hear clearly when I use the X-lite clients , but on Nokia E60 , I cannot hear anything ,ie whenever a call is made , the user who uses X-lite hears everything what the Nokia user says , but Nokai user cannot hear anything at all Please advice me , where I should check , the problem , is it because of codec selection , I did try with other codecs like ulaw , the experience was same I am using asterisk 1.2.8 on RHEL4 Thanks Joseph John my sip.conf contains [666] ; Xlite Phone username=666 type=friend secret=666 ;qualify=no ;port=5060 ;notransfer=yes host=dynamic context=from-internal disallow=all allow=alaw [221] ;; Nokia E-60 username=221 type=friend secret=221 ;qualify=no ;port=5060 ;notransfer=yes host=dynamic context=from-internal disallow=all allow=alaw Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite
Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work. That means, that you cant just walk down to your local airport, and use the IP part of the phone on their network. You have to have a non nat local server, to get it to run. Other than that, the phone can accept calls both from cellular network and IP network, and actuatly works quite well, both for cellular and IP traffic. But you cant do seamless handover, for example when you walk out of the building. You have two different numbers, your mobile number and your IP number And these cant automaticly be transferred. Hope this answeres your question Regards Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Olivier Krief Sendt: 7. juni 2006 16:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite 2006/6/7, Jon Schøpzinsky [EMAIL PROTECTED]: Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon What do you mean by users has to have some local equipment from the telco ? Do you think Nokia E60, E61 and E70 are appropriate for Fixed Mobile Convergence (each mobile phone being reachable at the same time from inhouse PBX and Telco's mobile network without any handover or roaming between both networks) ? Regards -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
