[asterisk-users] Set caller ID to anonymous
Hi guys, I am trying to set the caller ID to 'Anonymous anonymous' if the caller is not registered to the asterisk server. But I can't find a solution. Any ideas? Regards Philipp -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a ___ -- 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] CDR Rewrite -- Questions to the users
Ok, now for my long mail... tir, 13 01 2009 kl. 09:05 -0700, skrev Steve Murphy: CDR1: A - B start: e1a ans: e2 end: e4 Party: B disp: ANSW linkedID: abc9 CDR2: A start: e1 ans: e1 end: e6 Party: A disp: ANSW linkedID: abc9 Are start time and answer time the same in CDR2? CDR3: B - C start: e4 ans: e5 end: e6 Party: C disp: ANSW linkedid: abc9 CDR2 covers A (see the Party field), CDR1 covers B, CDR3 covers C. A's CDR could be used to bill A for his call in. It covers both the time A spent talking to B, and C. If you charge a different rate for A talking to B vs C, then you have some interesting SQL queries to make, I'll guess... As far as I'm concerned, A called B and that's what they pay for. The fact that B transferred them to a cell phone in Antarctica isn't A's problem, it's B's problem, so I'm happy with that. I need to somehow generate these CDR's to pass to our billing system: A-B start: e1 ans: e2 end: e6 disp: ANSW B-C start: e4 ans: e5 end: e6 disp: ANSW I can get that by looking at all foo-bar CDR's (CDR1) and look for a CDR's with the same linkedID and only foo (CDR2). Then I replace start and end times in CDR1 with the start and end times from CDR2, and I'm done. Nothing happens for CDR3 with this process, so I'm done. C's CDR records that B called C. It doesn't mention that A is doing all the talking. Perfect. B's CDR records the call from A to B; this is the only one that seems a little useless... It isn't useless, CDR2 is the one I need to get B as well as e2. Is this enough? If this is all you had, could you make it work? If you can't, would adding a field or two help? I am fairly certain it would be fine. Then the A does the transfer version... In the SImple CDR world, here's what would be produced: CDR1: A start: e1 ans: e1 end: e4 Party: A disp: ANSW linkedID: abc9 CDR2: A - B start: e1a ans: e2 end: e6 Party: B disp: ANSW linkedID: abc9 CDR3: A - C start: e4 ans: e5 end: e6 Party: C disp: ANSW linkedid: abc9 Here, A's total connection time is in CDR1; B with CDR2; C with CDR3. This is tricky... I need to create these CDR's for the billing system: src: A start: e1 ans: e2 end: e6 dst: B disp: ANSW src: A start: e4 ans: e5 end: e6 dst: B disp: ANSW If I do the same substitution again, I get this: A-B start: e1 ans: e2 end: e4. Whoops, end time is wrong. A-C start: e1 ans: e5 end: e4. Whoops, both start and end times are wrong. CDR2 needs to find e1 so it can replace start, while CDR3 shouldn't have anything replaced. I can't think of a query which will do this correctly. Again, is there enough info here for you to do what you need to do? If not what addition could be added to make it work? As far as I can tell, I won't be able to bill correctly for transfers with these CDR's. That isn't a regression by the way, so it shouldn't necessarily stop the switch to Simple CDR's. In the CDRfix2 doc, I outlined both the above blindxfer cases, and also permutations of attended xfers. Look them over, and see if what you need is possible with this format. The CDRfix2 doc is concerned with Leg-based CDR's. I haven't looked at those in-depth yet, because your proposal is to implement the Simple system first. /Benny ___ -- 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] mISDN BRI Asterisk 1.4
--- On Wed, 14/1/09, Ex Vito ex.vitor...@gmail.com wrote: While I don't know the OpenVOX B200P specifics, some interface cards need you to change physical jumpers in order to acheive NT vs TE, mode. Could that be the case ? -- exvito I've just checked the card and you were right the jumpers had not been changed to NT. I've done this now and also enabled power on one of the ports as this is also mentioned in the manual. However, still neither port comes up on L1 when I connect the router. Once I've changed this jumper setting do I still need to manually change the mISDN.conf file to use NT? When I do mISDN scan/config it is still setting the ports to TE which I then manually edit back to NT. Also, I guess at this point it doesn't matter for L1, but should I be using Point-To-Point or Point-To-Multipoint? Thanks ___ -- 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] Any free video (or audio) softphone VOIP client under Linux with touchscreen friendly interface ?
Hi, I'm curious if anyone knows of any possibility to use video VOIP client (like Ekiga or Linphone or...) under Linux that could be operated by touchscreen friendly GUI (bigger buttons, large keypad, etc...) ? I like Ekiga, but GUI is small and cannot be operated via touchscreen... But maybe there are some skins for existing clients that are more touchscreen friendly ? Thanks in advance, regards, Rob. ___ -- 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] gxp2000 and no sound asterisk 1.6
Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- 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] gxp2000 and no sound asterisk 1.6
On Wed, 14 Jan 2009, Ralf Träskman wrote: Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? First, upgrade your asterisk to 1.2 ... ;-) What is the external connection? Is it VoIP, PSTN, or ... ? If it's VoIP then it's almost certian to be a NAT problem with your network/router. There's no magic in setting up GXP2000's - they're fairly straightforward, and if you can do phone to phone, (via an asterisk) they're probably OK. Let us know more about the external connection technology... Gordon___ -- 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] Any free video (or audio) softphone VOIP client under Linux with touchscreen friendly interface ?
Twinkle has big buttons but it hasnt keypad to dial, the keypad is there to send DTMF. 2009/1/14 Robert Rozman robert.roz...@comutel.si Hi, I'm curious if anyone knows of any possibility to use video VOIP client (like Ekiga or Linphone or...) under Linux that could be operated by touchscreen friendly GUI (bigger buttons, large keypad, etc...) ? I like Ekiga, but GUI is small and cannot be operated via touchscreen... But maybe there are some skins for existing clients that are more touchscreen friendly ? Thanks in advance, regards, Rob. ___ -- 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 is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Set caller ID to anonymous
philipp-chemn...@gmx.de writes: I am trying to set the caller ID to 'Anonymous anonymous' if the caller is not registered to the asterisk server. But I can't find a solution. Which bit is causing you trouble? Detecting that the caller isn't registered, or setting caller ID? The latter is easy, just exten = _X!,n,Set(CALLERID(all)=Anonymous anonymous) /Benny ___ -- 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] Set caller ID to anonymous
On Wed, 14 Jan 2009 09:24:05 +0100, philipp-chemn...@gmx.de wrote: Hi guys, I am trying to set the caller ID to 'Anonymous anonymous' if the caller is not registered to the asterisk server. But I can't find a solution. put registered users in one context which dials out, and unregistered users in another which sets the callerid and then dials out. -- Regards, /\_/\ All dogs go to heaven. din...@alphaque.com(0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- 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] FWD and Asterisk
David @ULC schrieb: If I use below code in my sip.conf , [123] type=peer qualify=no port=5060 nat=no insecure=very this is very important host=voiper.ipkall.com dtmfmode=rfc2833 context=from-pstn canreinvite=no how will call understand that where I have to land as we DO NOT provide our IP in fwd configuration when we create an account. You, well, Asterisk on your behalf, registers with them and tells them your IP address. That's where inbound calls go. http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD http://www.voip-info.org/wiki-IPKall Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gxp2000 and no sound asterisk 1.6
Hi Yes we use voip as external. /ralf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: den 14 januari 2009 10:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] gxp2000 and no sound asterisk 1.6 On Wed, 14 Jan 2009, Ralf Träskman wrote: Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? First, upgrade your asterisk to 1.2 ... ;-) What is the external connection? Is it VoIP, PSTN, or ... ? If it's VoIP then it's almost certian to be a NAT problem with your network/router. There's no magic in setting up GXP2000's - they're fairly straightforward, and if you can do phone to phone, (via an asterisk) they're probably OK. Let us know more about the external connection technology... Gordon ___ -- 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] FWD and Asterisk
this is like the bible http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf 2009/1/14 Philipp Kempgen philipp.kemp...@amooma.de David @ULC schrieb: If I use below code in my sip.conf , [123] type=peer qualify=no port=5060 nat=no insecure=very this is very important host=voiper.ipkall.com dtmfmode=rfc2833 context=from-pstn canreinvite=no how will call understand that where I have to land as we DO NOT provide our IP in fwd configuration when we create an account. You, well, Asterisk on your behalf, registers with them and tells them your IP address. That's where inbound calls go. http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD http://www.voip-info.org/wiki-IPKall Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] evaluate SIP response codes in dialplan
Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken, error, circuit busy (e.g. 503) 486 is mapped to DIALSTATUS=BUSY but both 503 and 404 is mapped to DIALSTATUS=CONGESTION As when Asterisk forwards the response with SIP to the caller the same response code is used, I suspect this information must be stored somewhere inside the channel variable. So, are there any means to access it? thanks klaus ___ -- 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] gxp2000 and no sound asterisk 1.6
On Wed, 14 Jan 2009, Ralf Träskman wrote: Hi Yes we use voip as external. If the asterisk box is behind NAT itself, then you need to port-forward ports 5060 and 1-2 on the firewall to the asterisk box. Then you need to make sure that localnet= and externip= are set correctly in sip.conf. Gordon /ralf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: den 14 januari 2009 10:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] gxp2000 and no sound asterisk 1.6 On Wed, 14 Jan 2009, Ralf Träskman wrote: Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? First, upgrade your asterisk to 1.2 ... ;-) What is the external connection? Is it VoIP, PSTN, or ... ? If it's VoIP then it's almost certian to be a NAT problem with your network/router. There's no magic in setting up GXP2000's - they're fairly straightforward, and if you can do phone to phone, (via an asterisk) they're probably OK. Let us know more about the external connection technology... Gordon ___ -- 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] 0800 UK number
Why not send the closure reason in the text message? It costs the same to send 30 characters as it does 160. You also need to consider trunk capacity. if you send the message out it is likely that people will react immediately and call the number as soon as they receive the message. Depending on the number of people receiving the message you could get a lot of people getting busy when trying to retrieve the additional info. This may have a negative effect on the users perception of the system. Fadge -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 13 January 2009 18:30 To: David fire Cc: Asterisk Users Mailing List - Non-Commercial Discussion; aster...@dotr.com Subject: Re: [asterisk-users] 0800 UK number The number will also be used as a information line where a more detailed message can be played. The scenario is: 1) School is closed because the boiler has broken down. 2) The Head (or any authorised person) calls the service and leaves a detailed message of the reason for closure 3) The system sends the text message* to all subscribers saying School is closed today, please call foo for more details 4) If anyone wants more details, then they call in. * this message can be changed at any time by the Head texting it to the service. Julian David fire wrote: why 0800? the parents will subscribe to the system only once you have a lot of flat fee services on-line to call land lines/mobiles in UK. David 2009/1/13 Julian Lyndon-Smith aster...@dotr.com mailto:aster...@dotr.com I have concocted a system for my children's primary school where parents can dial in and subscribe to an emergency broadcast message so that they can be automatically contacted in case of a problem like the school being shut because of snow etc. I would like to provide an 0800 number service for this, so that there is no cost to the parents, but obviously I would like to get the best package possible. I have come across several packages, but would like the most inclusive minutes for the best price ;) Does anyone that has used an 0800 service in the UK have any recomendations ? Thanks Julian. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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 is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] mISDN BRI Asterisk 1.4
Lee Wilson wrote: --- On Wed, 14/1/09, Ex Vito ex.vitor...@gmail.com wrote: While I don't know the OpenVOX B200P specifics, some interface cards need you to change physical jumpers in order to acheive NT vs TE, mode. Could that be the case ? -- exvito I've just checked the card and you were right the jumpers had not been changed to NT. I've done this now and also enabled power on one of the ports as this is also mentioned in the manual. However, still neither port comes up on L1 when I connect the router. Once I've changed this jumper setting do I still need to manually change the mISDN.conf file to use NT? When I do mISDN scan/config it is still setting the ports to TE which I then manually edit back to NT. Also, I guess at this point it doesn't matter for L1, but should I be using Point-To-Point or Point-To-Multipoint? Thanks Yes, you would still need to configure mISDN correctly as well! And AFAIK you will need to use PTMP, as that is what the router would expect... -- Francesco Peeters Ubuntu all the way! 1 laptop, 1 server, 1 desktop at home and several servers in different locations ___ -- 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] evaluate SIP response codes in dialplan
Klaus Darilion schrieb: Is it somehow possible to evaluate the SIP response code inside the dialplan? No. Part of the reasoning is that Asterisk is meant to be a multi- protocol PBX, not a SIP softswitch. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] evaluate SIP response codes in dialplan
Philipp Kempgen schrieb: Klaus Darilion schrieb: Is it somehow possible to evaluate the SIP response code inside the dialplan? No. But if I remember correctly I have seen patches for that somewhere. Maybe on the bug tracker. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 0800 UK number
Hi Fadge thanks for the comments (see inline) asterisk wrote: Why not send the closure reason in the text message? It costs the same to send 30 characters as it does 160. You also need to consider trunk capacity. We had a recent problem where the school was closed because of a burst pipe. It took three days to fix - not the burst pipe, but because there was a suspicion that the pipe had asbestos around it, and the test took three days to come back. That is too much info for a text message ;) Normally, you are right - the message will be short and sweet so there will be no need to call in. if you send the message out it is likely that people will react immediately and call the number as soon as they receive the message. Depending on the number of people receiving the message you could get a lot of people getting busy when trying to retrieve the additional info. This may have a negative effect on the users perception of the system. We are putting it through a call center where we have 120 inbound lines. Given that few will call in if the message is suitably informative, then we should be able to cope. Fadge Thanks again. Julian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 13 January 2009 18:30 To: David fire Cc: Asterisk Users Mailing List - Non-Commercial Discussion; aster...@dotr.com Subject: Re: [asterisk-users] 0800 UK number The number will also be used as a information line where a more detailed message can be played. The scenario is: 1) School is closed because the boiler has broken down. 2) The Head (or any authorised person) calls the service and leaves a detailed message of the reason for closure 3) The system sends the text message* to all subscribers saying School is closed today, please call foo for more details 4) If anyone wants more details, then they call in. * this message can be changed at any time by the Head texting it to the service. Julian David fire wrote: why 0800? the parents will subscribe to the system only once you have a lot of flat fee services on-line to call land lines/mobiles in UK. David 2009/1/13 Julian Lyndon-Smith aster...@dotr.com mailto:aster...@dotr.com I have concocted a system for my children's primary school where parents can dial in and subscribe to an emergency broadcast message so that they can be automatically contacted in case of a problem like the school being shut because of snow etc. I would like to provide an 0800 number service for this, so that there is no cost to the parents, but obviously I would like to get the best package possible. I have come across several packages, but would like the most inclusive minutes for the best price ;) Does anyone that has used an 0800 service in the UK have any recomendations ? Thanks Julian. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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 is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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 email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] evaluate SIP response codes in dialplan
Philipp Kempgen schrieb: Philipp Kempgen schrieb: Klaus Darilion schrieb: Is it somehow possible to evaluate the SIP response code inside the dialplan? No. But if I remember correctly I have seen patches for that somewhere. Maybe on the bug tracker. http://bugs.digium.com/view.php?id=9743 But it doesn't have any patches attached. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] evaluate SIP response codes in dialplan
The simple answer is that Asterisk is too high-level. But you can change the response handlers in chan_sip.c to set various channel variables to achieve what you want pretty easily. Klaus Darilion wrote: Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken, error, circuit busy (e.g. 503) 486 is mapped to DIALSTATUS=BUSY but both 503 and 404 is mapped to DIALSTATUS=CONGESTION As when Asterisk forwards the response with SIP to the caller the same response code is used, I suspect this information must be stored somewhere inside the channel variable. So, are there any means to access it? thanks klaus ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] IAX Java Softphone?
I'm delighted to be able to say that as part of the agreement on my departure from Mexuar, the Corraleta applet source code Westhawk Ltd wrote for them has been released under the GPL. it is available for download at : http://www.mexuar.com/files/corraleta_sdk.rar Tim. On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ 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 ___ -- 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] evaluate SIP response codes in dialplan
Take a look (if it still exists) at the Asterisk B2BUA project. It has a patch that adds direct access to SIP response codes. It takes a little modification of the patch file to use in some of the newer asterisks (and to strip out the one codec option that's somewhat irrelevant), but it's a good starting spot. N. Klaus Darilion wrote: Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken, error, circuit busy (e.g. 503) 486 is mapped to DIALSTATUS=BUSY but both 503 and 404 is mapped to DIALSTATUS=CONGESTION As when Asterisk forwards the response with SIP to the caller the same response code is used, I suspect this information must be stored somewhere inside the channel variable. So, are there any means to access it? thanks klaus ___ -- 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] 0800 UK number
Why are you using a text message when you could be recording a message and sending it out? This would possibly be clearer than a read-and-callback scenario? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, January 14, 2009 8:11 AM To: asterisk Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'; aster...@dotr.com Subject: Re: [asterisk-users] 0800 UK number Hi Fadge thanks for the comments (see inline) asterisk wrote: Why not send the closure reason in the text message? It costs the same to send 30 characters as it does 160. You also need to consider trunk capacity. We had a recent problem where the school was closed because of a burst pipe. It took three days to fix - not the burst pipe, but because there was a suspicion that the pipe had asbestos around it, and the test took three days to come back. That is too much info for a text message ;) Normally, you are right - the message will be short and sweet so there will be no need to call in. if you send the message out it is likely that people will react immediately and call the number as soon as they receive the message. Depending on the number of people receiving the message you could get a lot of people getting busy when trying to retrieve the additional info. This may have a negative effect on the users perception of the system. We are putting it through a call center where we have 120 inbound lines. Given that few will call in if the message is suitably informative, then we should be able to cope. Fadge Thanks again. Julian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 13 January 2009 18:30 To: David fire Cc: Asterisk Users Mailing List - Non-Commercial Discussion; aster...@dotr.com Subject: Re: [asterisk-users] 0800 UK number The number will also be used as a information line where a more detailed message can be played. The scenario is: 1) School is closed because the boiler has broken down. 2) The Head (or any authorised person) calls the service and leaves a detailed message of the reason for closure 3) The system sends the text message* to all subscribers saying School is closed today, please call foo for more details 4) If anyone wants more details, then they call in. * this message can be changed at any time by the Head texting it to the service. Julian David fire wrote: why 0800? the parents will subscribe to the system only once you have a lot of flat fee services on-line to call land lines/mobiles in UK. David 2009/1/13 Julian Lyndon-Smith aster...@dotr.com mailto:aster...@dotr.com I have concocted a system for my children's primary school where parents can dial in and subscribe to an emergency broadcast message so that they can be automatically contacted in case of a problem like the school being shut because of snow etc. I would like to provide an 0800 number service for this, so that there is no cost to the parents, but obviously I would like to get the best package possible. I have come across several packages, but would like the most inclusive minutes for the best price ;) Does anyone that has used an 0800 service in the UK have any recomendations ? Thanks Julian. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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 is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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 email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email
Re: [asterisk-users] IAX Java Softphone?
Wow very cool - what is required for novices to install this application on their websites? Will you be making available some kind of easy install app? Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Wednesday, 14 January 2009 9:39 AM To: em...@mattruby.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Java Softphone? I'm delighted to be able to say that as part of the agreement on my departure from Mexuar, the Corraleta applet source code Westhawk Ltd wrote for them has been released under the GPL. it is available for download at : http://www.mexuar.com/files/corraleta_sdk.rar Tim. On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ 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 ___ -- 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] Set caller ID to anonymous
Hi, Am Mittwoch 14 Januar 2009 schrieb Benny Amorsen: philipp-chemn...@gmx.de writes: I am trying to set the caller ID to 'Anonymous anonymous' if the caller is not registered to the asterisk server. But I can't find a solution. Which bit is causing you trouble? Detecting that the caller isn't registered, or setting caller ID? The latter is easy, just exten = _X!,n,Set(CALLERID(all)=Anonymous anonymous) setting the caller ID works perfect. Detecting if a caller is or isn't registered is the problem. I'm using sip. Regards Philipp ___ -- 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] evaluate SIP response codes in dialplan
Philipp Kempgen schrieb: Klaus Darilion schrieb: Is it somehow possible to evaluate the SIP response code inside the dialplan? No. Part of the reasoning is that Asterisk is meant to be a multi- protocol PBX, not a SIP softswitch. This is IMO a stupid limitation. There are dozens of ISDN cause codes, dozens of SIP response codes and similar in other protocols, but Dial() only exports BUSY or CONGESTION .. thanks klaus ___ -- 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] IAX Java Softphone?
On Wed, 14 Jan 2009, Tim Panton wrote: I'm delighted to be able to say that as part of the agreement on my departure from Mexuar, the Corraleta applet source code Westhawk Ltd wrote for them has been released under the GPL. it is available for download at : http://www.mexuar.com/files/corraleta_sdk.rar Oooh, excellent! I had a play with this a year or so back and it worked very well. Shame my clients at the time didn't want it. Their loss! Now if only I knew how to install it... Cheers, Gordon ___ -- 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] evaluate SIP response codes in dialplan
- Klaus Darilion klaus.mailingli...@pernau.at wrote:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
Part of the reasoning is that Asterisk is meant to be a multi-
protocol PBX, not a SIP softswitch.
This is IMO a stupid limitation. There are dozens of ISDN cause codes,
dozens of SIP response codes and similar in other protocols, but
Dial()
only exports BUSY or CONGESTION ..
Right, app_dial condenses down the information it gets into some basic string
representations. You can also access a more specific Q.931 representation by
using the ${HANGUPCAUSE} dialplan variable. While this is not the SIP response
code this gives you more information. You can also control the SIP response
code by passing a Q.931 value to the Hangup() application itself. Unfortunately
the mappings of SIP response code - Q.931 are hard coded in chan_sip though
so that is where you can find what maps to what.
--
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] IAX Java Softphone?
Thank you for getting that code contributed to the community. Is there a spec somewhere of the features supported by that applet? A version history? Docs of the SDK it's distributed as? On Wed, 2009-01-14 at 14:38 +, Tim Panton wrote: I'm delighted to be able to say that as part of the agreement on my departure from Mexuar, the Corraleta applet source code Westhawk Ltd wrote for them has been released under the GPL. it is available for download at : http://www.mexuar.com/files/corraleta_sdk.rar Tim. On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein -- (C) Matthew Rubenstein ___ -- 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] IAX Java Softphone?
It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. On 14 Jan 2009, at 15:09, Dean Collins wrote: Wow very cool - what is required for novices to install this application on their websites? Will you be making available some kind of easy install app? Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Wednesday, 14 January 2009 9:39 AM To: em...@mattruby.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Java Softphone? I'm delighted to be able to say that as part of the agreement on my departure from Mexuar, the Corraleta applet source code Westhawk Ltd wrote for them has been released under the GPL. it is available for download at : http://www.mexuar.com/files/corraleta_sdk.rar Tim. On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ 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 ___ -- 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] IAX Java Softphone?
Since we are all learners here, you can download the Java stuff for free from sun, but you'd need about as much time on the Java as you spend on *. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Wednesday, January 14, 2009 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Java Softphone? It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. On 14 Jan 2009, at 15:09, Dean Collins wrote: Wow very cool - what is required for novices to install this application on their websites? Will you be making available some kind of easy install app? Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Wednesday, 14 January 2009 9:39 AM To: em...@mattruby.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Java Softphone? I'm delighted to be able to say that as part of the agreement on my departure from Mexuar, the Corraleta applet source code Westhawk Ltd wrote for them has been released under the GPL. it is available for download at : http://www.mexuar.com/files/corraleta_sdk.rar Tim. On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ 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 ___ -- 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] Upgrade to v.1.2.31 ... weird change
Tilghman Lesher escreveu: On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote: I think it happened when I upgraded an install to 1.2.31 The variable CALLERIDNUM no longer works and CallerID(num) has to be used. I don't see why not. There has been no change whatsoever to that body of code. I think there is some mistake on his test about CALLERIDNUM. Did a quick test here on 1.2.31 and it's working fine. On the other hand, the change in chan_iax2.c on 1.2.31 really broke something... but it's not related to dialplan variables. IAX2 peer registration: http://bugs.digium.com/view.php?id=14238 Leonardo ___ -- 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] Upgrade to v.1.2.31 ... weird change
On Wed, Jan 14, 2009 at 02:56:44PM -0200, Leonardo Gomes Figueira wrote:
Tilghman Lesher escreveu:
On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote:
I think it happened when I upgraded an install to 1.2.31
The variable CALLERIDNUM no longer works and CallerID(num) has to be
used.
I don't see why not. There has been no change whatsoever to that body of
code.
I think there is some mistake on his test about CALLERIDNUM. Did a quick
test here on 1.2.31 and it's working fine.
I'll check, but something definately changed. I think I was using
${CALLERIDNUM:1:4} anyway didn't work as planned. Changing to
CallerID(num) and it worked again.
Steve
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Re: [asterisk-users] IAX Java Softphone?
Tim Panton ha scritto: It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. Really great stuff! Could you please explain how to use it in a java application? Thanks in advance. On 14 Jan 2009, at 15:09, Dean Collins wrote: Wow very cool - what is required for novices to install this application on their websites? Will you be making available some kind of easy install app? Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Wednesday, 14 January 2009 9:39 AM To: em...@mattruby.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Java Softphone? I'm delighted to be able to say that as part of the agreement on my departure from Mexuar, the Corraleta applet source code Westhawk Ltd wrote for them has been released under the GPL. it is available for download at : http://www.mexuar.com/files/corraleta_sdk.rar Tim. On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ 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 ___ -- 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
[asterisk-users] G.729.1 - any interest?
The G.729.1 wideband codec is starting to show a slight bit of traction. There is a possibility that Asterisk could support G.729.1 - would you use it or buy it if it was available? More importantly, does any equipment with which your systems currently exchange traffic support G.729.1? Currently, the number of devices supporting G.729.1 seems to be fairly limited and it may be an imbalanced decision to support a codec that nobody else uses. If G.729.1 were to be offered as a codec for Asterisk by Digium, it would have to be as a commercial product, as the codec is patent- encumbered. Pricing and licensing terms are outside the scope of this discussion, but I would expect something like G.729. Of course, passthrough-mode (non-transcoding) would not require licensing with Asterisk and is outside of the scope of this question. Timing is also an unknown issue - there are obviously many other projects in the pipeline for the Digium engineering team to work on before this probably could be completed, even if the decision is made to pursue a development effort. Note that G.722 is free and already available, and may have similar MOS scores (but certainly not exactly similar) as that of G.729.1. Comparisons of G.729.1 and G.722 are left as exercises to the reader, or see the excellent presentation below which is quite enlightening. Your opinions are welcome on the topic! Resources: http://portal.etsi.org/stq/workshop2007presentations/quinquis_slides.pdf http://en.wikipedia.org/wiki/G.729.1 http://en.wikipedia.org/wiki/G.722 [Apologies for the cross-post - this has some interest to both the user and development community, I think. I'll also apologize for what is a post about issues that are not open-source, but it seems that within Digium I'm probably the most appropriate person to canvass the community on this particular question, as it involves gauging the general thinking of the VoIP community and is not merely a Digium-only concern.] JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- 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] IAX Java Softphone?
hi thanks for this code is a very good contribution is there any demo? example? or how to? or any docs? thanks David 2009/1/14 Roberto Fichera ker...@tekno-soft.it Tim Panton ha scritto: It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. Really great stuff! Could you please explain how to use it in a java application? Thanks in advance. On 14 Jan 2009, at 15:09, Dean Collins wrote: Wow very cool - what is required for novices to install this application on their websites? Will you be making available some kind of easy install app? Regards, Dean Collins Cognation incd...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Wednesday, 14 January 2009 9:39 AM To: em...@mattruby.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Java Softphone? I'm delighted to be able to say that as part of the agreement on my departure from Mexuar, the Corraleta applet source code Westhawk Ltd wrote for them has been released under the GPL. it is available for download at : http://www.mexuar.com/files/corraleta_sdk.rar Tim. On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ 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 ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] G.729.1 - any interest?
John Todd wrote: The G.729.1 wideband codec is starting to show a slight bit of traction. There is a possibility that Asterisk could support G.729.1 - would you use it or buy it if it was available? More importantly, does any equipment with which your systems currently exchange traffic support G.729.1? Currently, the number of devices supporting G.729.1 seems to be fairly limited and it may be an imbalanced decision to support a codec that nobody else uses. If G.729.1 were to be offered as a codec for Asterisk by Digium, it would have to be as a commercial product, as the codec is patent- encumbered. Pricing and licensing terms are outside the scope of this discussion, but I would expect something like G.729. Of course, passthrough-mode (non-transcoding) would not require licensing with Asterisk and is outside of the scope of this question. Timing is also an unknown issue - there are obviously many other projects in the pipeline for the Digium engineering team to work on before this probably could be completed, even if the decision is made to pursue a development effort. Note that G.722 is free and already available, and may have similar MOS scores (but certainly not exactly similar) as that of G.729.1. Comparisons of G.729.1 and G.722 are left as exercises to the reader, or see the excellent presentation below which is quite enlightening. Your opinions are welcome on the topic! Resources: http://portal.etsi.org/stq/workshop2007presentations/quinquis_slides.pdf http://en.wikipedia.org/wiki/G.729.1 http://en.wikipedia.org/wiki/G.722 [Apologies for the cross-post - this has some interest to both the user and development community, I think. I'll also apologize for what is a post about issues that are not open-source, but it seems that within Digium I'm probably the most appropriate person to canvass the community on this particular question, as it involves gauging the general thinking of the VoIP community and is not merely a Digium-only concern.] Where have you seen it getting traction? France Telecom came up with it, and are using it, but that's kind of isolated from the rest of the universe. The PDF you referenced is little more than a France Telecoms sales pitch for G.729.1. Audiocodes announced something, but its vague and they aren't shipping yet. AMR-WB would make more sense, as 3G cellphones all use it, and transcoding these things looses huge amounts of quality. G.722.1 is also getting somewhere, largely because of Polycom's commitment to it. The really wacky one is G.711.1. Has anyone heard of people taking that seriously. Regards, Steve ___ -- 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] 0800 UK number
Danny Nicholas wrote: Why are you using a text message when you could be recording a message and sending it out? This would possibly be clearer than a read-and-callback scenario? Do you think so? Remembering that most people, if they pick up the phone to hear a recorded message will immediately hang up without listening to it, then there's the few that will listen and not get it or understand, either because they have answered their phone in a crowded room expecting a real person to be on the end or because a lot of the time you don't hear a recorded message the first time you listen to it (even if you're expecting it to be recorded). I think a text message is a much more elegant way (presumably relayed though an SMSC so that sending the message to all users doesn't take a day to do). ___ -- 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] IAX Java Softphone?
On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote: Thank you for getting that code contributed to the community. Is there a spec somewhere of the features supported by that applet? A version history? Docs of the SDK it's distributed as? All I have is the link. I should emphasise that I no longer have any relationship with Mexuar so I'm in the dark as to exactly what their plans are as far as supporting this code is concerned. I'm just one of the original authors and an open-source proponent. I guess it would make sense for someone to open a sourceforge project for it and add those things. Tim. ___ -- 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] IAX Java Softphone?
On 14 Jan 2009, at 17:07, Roberto Fichera wrote: Tim Panton ha scritto: It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. Really great stuff! Could you please explain how to use it in a java application? Thanks in advance. I designed it as a Java applet, so the top level needs Javascript and DHTML from the browser to provide a UI. That said, It wouldn't be very hard to write an application class and some UI classes to turn it into a stand-alone application , but that depends on the complexity of the UI you want. ___ -- 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] 0800 UK number
On Wed, 14 Jan 2009, Thomas Kenyon wrote: Danny Nicholas wrote: Why are you using a text message when you could be recording a message and sending it out? This would possibly be clearer than a read-and-callback scenario? Do you think so? Remembering that most people, if they pick up the phone to hear a recorded message will immediately hang up without listening to it, then there's the few that will listen and not get it or understand, either because they have answered their phone in a crowded room expecting a real person to be on the end or because a lot of the time you don't hear a recorded message the first time you listen to it (even if you're expecting it to be recorded). I think a text message is a much more elegant way (presumably relayed though an SMSC so that sending the message to all users doesn't take a day to do). It's about 5 seconds to send a message with a GSM terminal, so 20 minutes for 250... Which might be OK, depending on the number of messages required... (Although cost is another factor - for those not in the UK, it costs to send a text message, and it's free to receive) Gordon ___ -- 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] evaluate SIP response codes in dialplan
Klaus Darilion schrieb: Philipp Kempgen schrieb: Klaus Darilion schrieb: Is it somehow possible to evaluate the SIP response code inside the dialplan? No. Part of the reasoning is that Asterisk is meant to be a multi- protocol PBX, not a SIP softswitch. This is IMO a stupid limitation. There are dozens of ISDN cause codes, dozens of SIP response codes and similar in other protocols, but Dial() only exports BUSY or CONGESTION .. I know. But the developers didn't want to add it. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729.1 - any interest?
On Jan 14, 2009, at 12:27 PM, Steve Underwood wrote: John Todd wrote: The G.729.1 wideband codec is starting to show a slight bit of traction. There is a possibility that Asterisk could support G.729.1 - would you use it or buy it if it was available? More importantly, does any equipment with which your systems currently exchange traffic support G.729.1? Currently, the number of devices supporting G.729.1 seems to be fairly limited and it may be an imbalanced decision to support a codec that nobody else uses. If G.729.1 were to be offered as a codec for Asterisk by Digium, it would have to be as a commercial product, as the codec is patent- encumbered. Pricing and licensing terms are outside the scope of this discussion, but I would expect something like G.729. Of course, passthrough-mode (non-transcoding) would not require licensing with Asterisk and is outside of the scope of this question. Timing is also an unknown issue - there are obviously many other projects in the pipeline for the Digium engineering team to work on before this probably could be completed, even if the decision is made to pursue a development effort. Note that G.722 is free and already available, and may have similar MOS scores (but certainly not exactly similar) as that of G.729.1. Comparisons of G.729.1 and G.722 are left as exercises to the reader, or see the excellent presentation below which is quite enlightening. Your opinions are welcome on the topic! Resources: http://portal.etsi.org/stq/workshop2007presentations/quinquis_slides.pdf http://en.wikipedia.org/wiki/G.729.1 http://en.wikipedia.org/wiki/G.722 [Apologies for the cross-post - this has some interest to both the user and development community, I think. I'll also apologize for what is a post about issues that are not open-source, but it seems that within Digium I'm probably the most appropriate person to canvass the community on this particular question, as it involves gauging the general thinking of the VoIP community and is not merely a Digium- only concern.] Where have you seen it getting traction? France Telecom came up with it, and are using it, but that's kind of isolated from the rest of the universe. The PDF you referenced is little more than a France Telecoms sales pitch for G.729.1. Audiocodes announced something, but its vague and they aren't shipping yet. AMR-WB would make more sense, as 3G cellphones all use it, and transcoding these things looses huge amounts of quality. G.722.1 is also getting somewhere, largely because of Polycom's commitment to it. The really wacky one is G.711.1. Has anyone heard of people taking that seriously. Regards, Steve slight bit = Audiocodes, and SPIRIT DSP code on some TI chips. Others? I don't know, I'd be interested in seeing if so. G.711.1 is still a ghost codec, from what I've been able to see. JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- 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] IAX Java Softphone?
Tim Panton ha scritto: On 14 Jan 2009, at 17:07, Roberto Fichera wrote: Tim Panton ha scritto: It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. Really great stuff! Could you please explain how to use it in a java application? Thanks in advance. I designed it as a Java applet, so the top level needs Javascript and DHTML from the browser to provide a UI. That said, It wouldn't be very hard to write an application class and some UI classes to turn it into a stand-alone application , but that depends on the complexity of the UI you want. I'm interested to use it as IAX2 API within my UI, so something like: - open IAX2 channel - call 123456 - answer a call - close IAX2 channel ___ -- 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] agi and set variable ( accountcode in aserisk 1.4)
i am set var Set(CDR(accountcode)=forkcdr-test) into agiphp
probe
$agi-exec('Set(CDR(accountcode)=5)');
$agi-exec('SetAccount','123123123');
and no work ...
how to solutions.
thanks people!
___
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Re: [asterisk-users] IAX Java Softphone?
On 14 Jan 2009, at 18:02, Roberto Fichera wrote: Tim Panton ha scritto: On 14 Jan 2009, at 17:07, Roberto Fichera wrote: Tim Panton ha scritto: It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. Really great stuff! Could you please explain how to use it in a java application? Thanks in advance. I designed it as a Java applet, so the top level needs Javascript and DHTML from the browser to provide a UI. That said, It wouldn't be very hard to write an application class and some UI classes to turn it into a stand-alone application , but that depends on the complexity of the UI you want. I'm interested to use it as IAX2 API within my UI, so something like: - open IAX2 channel - call 123456 - answer a call - close IAX2 channel It is definitely capable of that with an added class or 2. - but remember it is GPL, so you would 'taint' the rest of your code - if it isn't already GPL. - Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] IAX Java Softphone?
On Wed, 2009-01-14 at 17:38 +, Tim Panton wrote: On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote: Thank you for getting that code contributed to the community. Is there a spec somewhere of the features supported by that applet? A version history? Docs of the SDK it's distributed as? All I have is the link. I should emphasise that I no longer have any relationship with Mexuar so I'm in the dark as to exactly what their plans are as far as supporting this code is concerned. I'm just one of the original authors and an open-source proponent. I guess it would make sense for someone to open a sourceforge project for it and add those things. Do you know if there are at least hooks in there for the applet to do video over IAX? Tim. -- (C) Matthew Rubenstein ___ -- 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] IAX Java Softphone?
On 14 Jan 2009, at 18:11, Matthew Rubenstein wrote: On Wed, 2009-01-14 at 17:38 +, Tim Panton wrote: On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote: Thank you for getting that code contributed to the community. Is there a spec somewhere of the features supported by that applet? A version history? Docs of the SDK it's distributed as? All I have is the link. I should emphasise that I no longer have any relationship with Mexuar so I'm in the dark as to exactly what their plans are as far as supporting this code is concerned. I'm just one of the original authors and an open-source proponent. I guess it would make sense for someone to open a sourceforge project for it and add those things. Do you know if there are at least hooks in there for the applet to do video over IAX? No, there aren't. We didn't even implement the video frame classes. I don't think it would be hard to add support for a simple video codec transport. The problem is the renderer. Java basically doesn't promise to deliver any video codecs. You are at the mercy of what happens to be installed on the OS or by 3rd parties (eg Quicktime, DiVX etc). (Caveat - I haven't investigated this for a while, it may be that JavaFX changes this picture) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] Strange IAX2 registration issue
I have a single connection that seems to register ok but then becomes unregistered immediately. This is what I see with IAX debug turned on: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 6ms SCall: 1 DCall: 0 [76.25.248.23:4569] USERNAME: ashlawn-cfam REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGACK Timestamp: 2ms SCall: 2 DCall: 1 [76.25.248.23:4569] USERNAME: ashlawn-cfam DATE TIME : 2009-01-14 10:36:20 REFRESH : 60 APPARENT ADDRES : IPV4 204.144.134.114:1047 Here is what i have in iax.conf for this connection: [ashlawn-cfam] type=friend context= host=dynamic secret= disallow=all allow=gsm allow=ulaw The weird part is that port 1047. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 v...@rockynet.com ___ -- 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] IAX Java Softphone?
Tim Panton ha scritto: On 14 Jan 2009, at 18:02, Roberto Fichera wrote: Tim Panton ha scritto: On 14 Jan 2009, at 17:07, Roberto Fichera wrote: Tim Panton ha scritto: It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. Really great stuff! Could you please explain how to use it in a java application? Thanks in advance. I designed it as a Java applet, so the top level needs Javascript and DHTML from the browser to provide a UI. That said, It wouldn't be very hard to write an application class and some UI classes to turn it into a stand-alone application , but that depends on the complexity of the UI you want. I'm interested to use it as IAX2 API within my UI, so something like: - open IAX2 channel - call 123456 - answer a call - close IAX2 channel It is definitely capable of that with an added class or 2. Could you point me in the proper source code so I can have a look in? - but remember it is GPL, so you would 'taint' the rest of your code - if it isn't already GPL. I generally follow the rule than if the library is GPL and if the end user ask for the source code I'll provide the source code as it should. If I made some changes in the GPL code, it will be always released to the original author. In all cases the GPL libraries are always mentioned as they are in our custom applications. We generally use jfreechart, jasper report and so on in our applications with this rules. Wouldn't be sufficient for you ;-)? - Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms?
Hi, I've been noticing a lot of these messages lately: NOTICE[10235]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? Is something broken? I'm running asterisk-1.4.22.1. They seem to happen in a number of different places where a beep or recording is played, such as when someone leaves voicemail or when an AGI script I have plays a time announcement -- lots of different places. . -- Executing [...@main-menu:2] Wait(SIP/redacted-09501e28, 1) in new stack -- Executing [...@main-menu:3] VoiceMail(SIP/redacted-09501e28, 10|s) in new stack -- SIP/redacted-09501e28 Playing 'beep' (language 'en') [Jan 14 12:34:18] NOTICE[10030]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/10/tmp/EGIKgI format: wav49, 0x9489080 -- x=1, open writing: /var/spool/asterisk/voicemail/default/10/tmp/EGIKgI format: gsm, 0x94feaf8 -- x=2, open writing: /var/spool/asterisk/voicemail/default/10/tmp/EGIKgI format: wav, 0x950c6e0 -- User hung up . . -- Executing [...@main-menu:3] VoiceMail(SIP/redacted-09500428, 10|s) in new stack -- SIP/redacted-09500428 Playing 'beep' (language 'en') [Jan 14 13:23:05] NOTICE[10235]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/10/tmp/o9SnHz format: wav49, 0x94feaf8 -- x=1, open writing: /var/spool/asterisk/voicemail/default/10/tmp/o9SnHz format: gsm, 0x9489080 -- x=2, open writing: /var/spool/asterisk/voicemail/default/10/tmp/o9SnHz format: wav, 0x9523bf8 -- User hung up . -- Launched AGI Script /var/lib/asterisk/agi-bin/talking-clock.agi -- AGI Script Executing Application: (PlayTones) Options: (!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!415/1500) -- Playing 'at-tone-time-exactly' (escape_digits=) (sample_offset 0) [Jan 14 13:35:07] NOTICE[10271]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? -- Zap/5-1 Playing 'digits/1' (language 'en') -- Zap/5-1 Playing 'digits/30' (language 'en') [Jan 14 13:35:08] NOTICE[10271]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? -- Zap/5-1 Playing 'digits/5' (language 'en') [Jan 14 13:35:09] NOTICE[10271]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? -- Zap/5-1 Playing 'vm-and' (language 'en') -- Zap/5-1 Playing 'digits/15' (language 'en') -- Zap/5-1 Playing 'seconds' (language 'en') -- AGI Script Executing Application: (PlayTones) Options: (!0/500,!523/20,!0/980,!523/20,!0/980,!415/1500) == Spawn extension (from-pots1, s, 2) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' . -- Mark G. Thomas (m...@misty.com) ___ -- 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-dev] G.729.1 - any interest?
Stéphane Van Geystelen wrote: You know my opinion about it ;) The G729.1 is still a 50Hz 7000kHz bandwidth. An ultra wideband codec capabilities would be a real breakthrough. 7KHz is not ultra-wideband, it's wideband. There are already wideband codecs out there, including G.722, G.722.1 and AMR-WB, to name a few. There are also ultra-wideband codecs (16KHz bandwidth) including G.722.1C. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.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
Re: [asterisk-users] 0800 UK number
Gordon Henderson wrote: On Wed, 14 Jan 2009, Thomas Kenyon wrote: Danny Nicholas wrote: Why are you using a text message when you could be recording a message and sending it out? This would possibly be clearer than a read-and-callback scenario? The sms will be sufficient for most people (The school is closed today because of foo. Please call baa for more info), and only those who *really* want to know will call in. Do you think so? Remembering that most people, if they pick up the phone to hear a recorded message will immediately hang up without listening to it, then yup. there's the few that will listen and not get it or understand, either because they have answered their phone in a crowded room expecting a real person to be on the end or because a lot of the time you don't hear a recorded message the first time you listen to it (even if you're expecting it to be recorded). yup I think a text message is a much more elegant way (presumably relayed though an SMSC so that sending the message to all users doesn't take a day to do). It's about 5 seconds to send a message with a GSM terminal, so 20 minutes for 250... Which might be OK, depending on the number of messages required... (Although cost is another factor - for those not in the UK, it costs to send a text message, and it's free to receive) we send all of our sms via clickatell - very quick, half the price of sending it via the PRI Thanks to all for the help and advice. Julian Gordon ___ -- 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 email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] IAX Java Softphone?
Tim -
Do you have any minimal docs or hints on what hooks the DHTML/JS methods
are available for scripting? Something like a quickstart javascript example?
I'm great with javascript, but I havn't read thru the Java to figure out
the hooks yet - if thats whats needed, I dont mind, but I'd rather hear
from the guy who knows best.
I'm assuming something like:
applet id=xyz ...
script
var applet = [get applet ref];
function onDialButtonClick()
{
var number = myFunctionGetPhoneNumber();
applet.connectToServer(my.iax.server.com,user,pass);
applet.dial(number);
[update UI]
}
function onHangupClick() { applet.hangupCall();applet.disconnectServer() }
/script
Something like that?
-josiah
Tim Panton wrote:
On 14 Jan 2009, at 18:02, Roberto Fichera wrote:
Tim Panton ha scritto:
On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
Tim Panton ha scritto:
It isn't really in a state for novices at the present
you'd need:
1) a java compiler
2) a java code signing certificate (java applets can't read from
the
mic
without being signed)
3) appropriate javascript and DHTML to implement the look and feel
4) an asterisk (or freeSWITCH) to talk IAX to.
Tim.
Really great stuff! Could you please explain how to use it in a java
application?
Thanks in advance.
I designed it as a Java applet, so the top level needs Javascript and
DHTML from the
browser to provide a UI.
That said, It wouldn't be very hard to write an application class and
some
UI classes to turn it into a stand-alone application , but that
depends on the
complexity of the UI you want.
I'm interested to use it as IAX2 API within my UI, so something like:
- open IAX2 channel
- call 123456
- answer a call
- close IAX2 channel
It is definitely capable of that with an added class or 2.
- but remember it is GPL, so you would 'taint' the rest of your code
- if it isn't already GPL.
-
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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--
Josiah Bryan
IT Manager
Productive Concepts, Inc.
jbr...@productiveconcepts.com
(765) 964-6009, ext. 224
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Re: [asterisk-users] IAX Java Softphone?
On 14 Jan 2009, at 19:53, Josiah Bryan wrote:
Tim -
Do you have any minimal docs or hints on what hooks the DHTML/JS
methods
are available for scripting? Something like a quickstart javascript
example?
I'm great with javascript, but I havn't read thru the Java to figure
out
the hooks yet - if thats whats needed, I dont mind, but I'd rather
hear
from the guy who knows best.
I'm assuming something like:
applet id=xyz ...
script
var applet = [get applet ref];
function onDialButtonClick()
{
var number = myFunctionGetPhoneNumber();
applet.connectToServer(my.iax.server.com,user,pass);
applet.dial(number);
[update UI]
}
function onHangupClick()
{ applet.hangupCall();applet.disconnectServer() }
/script
Something like that?
-josiah
It's up to Mexuar to decide if they want to release any pre-existing
documentation
(and since it isn't in the .rar I guess they don't intend to at the
moment).
The easiest thing would be to run JavaDoc over the applet class and
see what public methods exist.
Tim.
--
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www.westhawk.co.uk
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Re: [asterisk-users] IAX Java Softphone?
On 14 Jan 2009, at 18:36, Roberto Fichera wrote: Tim Panton ha scritto: On 14 Jan 2009, at 18:02, Roberto Fichera wrote: Tim Panton ha scritto: On 14 Jan 2009, at 17:07, Roberto Fichera wrote: Tim Panton ha scritto: It isn't really in a state for novices at the present you'd need: 1) a java compiler 2) a java code signing certificate (java applets can't read from the mic without being signed) 3) appropriate javascript and DHTML to implement the look and feel 4) an asterisk (or freeSWITCH) to talk IAX to. Tim. Really great stuff! Could you please explain how to use it in a java application? Thanks in advance. I designed it as a Java applet, so the top level needs Javascript and DHTML from the browser to provide a UI. That said, It wouldn't be very hard to write an application class and some UI classes to turn it into a stand-alone application , but that depends on the complexity of the UI you want. I'm interested to use it as IAX2 API within my UI, so something like: - open IAX2 channel - call 123456 - answer a call - close IAX2 channel It is definitely capable of that with an added class or 2. Could you point me in the proper source code so I can have a look in? ./corraleta/protocol/netse/BinderSE.java Has a Main method used to test the protocol that would be a good place to start. - but remember it is GPL, so you would 'taint' the rest of your code - if it isn't already GPL. I generally follow the rule than if the library is GPL and if the end user ask for the source code I'll provide the source code as it should. If I made some changes in the GPL code, it will be always released to the original author. In all cases the GPL libraries are always mentioned as they are in our custom applications. We generally use jfreechart, jasper report and so on in our applications with this rules. Wouldn't be sufficient for you ;-)? Not my copyright - not my decision ;-) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] mISDN BRI Asterisk 1.4
Also, I guess at this point it doesn't matter for L1, but should I be using Point-To-Point or Point-To-Multipoint? Thanks Yes, you would still need to configure mISDN correctly as well! And AFAIK you will need to use PTMP, as that is what the router would expect... -- Francesco Peeters Thanks for clarifying I've double-checked that it is running ptmp but still no link lights. Anyone got other suggestions? Regards Lee ___ -- 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] IAX Java Softphone?
Tim Panton wrote:
On 14 Jan 2009, at 19:53, Josiah Bryan wrote:
Tim -
Do you have any minimal docs or hints on what hooks the DHTML/JS
methods
are available for scripting? Something like a quickstart javascript
example?
I'm great with javascript, but I havn't read thru the Java to figure
out
the hooks yet - if thats whats needed, I dont mind, but I'd rather
hear
from the guy who knows best.
I'm assuming something like:
applet id=xyz ...
script
var applet = [get applet ref];
function onDialButtonClick()
{
var number = myFunctionGetPhoneNumber();
applet.connectToServer(my.iax.server.com,user,pass);
applet.dial(number);
[update UI]
}
function onHangupClick()
{ applet.hangupCall();applet.disconnectServer() }
/script
Something like that?
-josiah
It's up to Mexuar to decide if they want to release any pre-existing
documentation
(and since it isn't in the .rar I guess they don't intend to at the
moment).
The easiest thing would be to run JavaDoc over the applet class and
see what public methods exist.
Understood - thanks for your patience with these questions.
Regards,
-josiah
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Re: [asterisk-users] 0800 UK number
On Wed, 14 Jan 2009, Julian Lyndon-Smith wrote: It's about 5 seconds to send a message with a GSM terminal, so 20 minutes for 250... Which might be OK, depending on the number of messages required... (Although cost is another factor - for those not in the UK, it costs to send a text message, and it's free to receive) we send all of our sms via clickatell - very quick, half the price of sending it via the PRI I looked at these people some time back for bulk SMSs for a project I was quoting on: (all UK based) http://www.smscarrier.com/ I won the contract for the project then they cancelled it. Ho hum! Gordon ___ -- 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] Nortel files for bankruptcy protection
Nortel filed for bankruptcy today -Karl A/P: http://www.google.com/hostednews/ap/article/ALeqM5gx8oAvO1SIb6Ya2KhA2d-d9SZunwD95N5HVG0 ___ -- 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] AMI API , Editing extensions.conf
Hello list,
I'm using a PHP script to communicate with asterisk via AMI, and edit
configuration files. So far everything went ok, but I came up with a
little problem editing extensions.conf using 'updateconfig'.
Is it possible to edit an existing line in extensions.conf file?, e.g.
Given a 'category' in my extensions.conf as follow:
[macro-mytest]
exten = s,1,Dial(${SYSTEM_TECHS}/${ar...@${system_sip_provider},,kKtTwW)
exten = s,n,Hangup()
I send this request to Asterisk:
...
Action: UpdateConfig
SrcFilename: extensions.conf
DstFilename: /tmp/extensions.conf
Action-00: Update
Cat-00: macro-TPLtrunkcell
Var-00: exten
Value-00: s,n,Dial(${SYSTEM_TECHZ}/${ARG1},,kKtTwW)
Match-00: SYSTEM_TECH
...
After the request, which Asterisk execute successfully, the result is
something like:
[macro-mytest]
exten = s,1,Dial(${SYSTEM_TECHS}/${ar...@${system_sip_provider},,kKtTwW)
exten = s,n,Hangup()
exten = s,n,Dial(${SYSTEM_TECHZ}/${ARG1},,kKtTwW)
Instead of :
[macro-mytest]
exten = s,1,Dial(${SYSTEM_TECHZ}/${ARG1},,kKtTwW)
exten = s,n,Hangup()
... which was the answer I was specting.
I'm I forgetting something here?
(In the source I saw an extra parameter to AMI when using the
UpdateConfig command, which was 'Line-XX', I tried indicating the
exact line too, but that did not help).
Thanks for your help. Best regards,
--
Jose P. Espinal
http://blog.Slackware-Es.com
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[asterisk-users] 1.6.1-b4: Can't get fax2mail work from System()
On 1.6.1-beta4:
Trying to receive faxes over a pstn line. extensions.conf:
[incoming-pstn-line]
exten = fax,1,NoOp(Fax Detected)
exten = fax,2,GoTo(incoming-fax,s,1)
exten = fax,n,Hangup()
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM})
exten = s,2,ReceiveFAX(${FAXFILE}.tif)
exten = s,3,Hangup()
exten=h,1,System(/usr/local/bin/fax2mail.1.sh --dest-name Sean
--dest-email ${Sean_email} -f ${FAXFILE})
which looks like it works just fine from the cli:
-- DAHDI/2-1 is ringing
-- Redirecting DAHDI/4-1 to fax extension
-- Hungup 'DAHDI/2-1'
== Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on
'DAHDI/4-1'
-- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax
Detected) in new stack
-- Executing [...@incoming-pstn-line:2] Goto(DAHDI/4-1,
incoming-fax,s,1) in new stack
-- Goto (incoming-fax,s,1)
-- Executing [...@incoming-fax:1] Set(DAHDI/4-1,
FAXFILE=/var/spool/asterisk/fax/200901141711-0) in new stack
-- Executing [...@incoming-fax:2] ReceiveFAX(DAHDI/4-1,
/var/spool/asterisk/fax/200901141711-0.tif) in new stack
-- Executing [...@incoming-fax:3] Hangup(DAHDI/4-1, ) in new stack
== Spawn extension (incoming-fax, s, 3) exited non-zero on 'DAHDI/4-1'
-- Executing [...@incoming-fax:1] System(DAHDI/4-1,
/usr/local/bin/fax2mail.1.sh --dest-name Sean --dest-email
seandar...@gmail.com -f /var/spool/asterisk/fax/200901141711-0) in
new stack
-- Hungup 'DAHDI/4-1'
But it doesn't - no email is ever sent. BUT, if I execute the fax2mail
cmd from the terminal (pasting from the cli output) it sends the email:
/usr/local/bin/fax2mail.1.sh --dest-name Sean --dest-email
seandar...@gmail.com -f /var/spool/asterisk/fax/200901141711-0
Am I screwing up the System() command somehow? Is System() screwed up in
1.6.1?
Any clues how to debug this? I did find one relevant thread
http://asteriskforum.ru/viewtopic.php?p=15629 , which is unfortunatley
in Russian. In that thread someone figured out how to turn on DEBUG for
app_fax. How did you do that?
sean
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Re: [asterisk-users] 1.6.1-b4: Can't get fax2mail work from System()
Start with your mail log. Any errors visible?
How about system log - PAMpermission errors?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: January 14, 2009 5:31 PM
To: Asterisk Users List
Subject: [asterisk-users] 1.6.1-b4: Can't get fax2mail work from System()
On 1.6.1-beta4:
Trying to receive faxes over a pstn line. extensions.conf:
[incoming-pstn-line]
exten = fax,1,NoOp(Fax Detected)
exten = fax,2,GoTo(incoming-fax,s,1)
exten = fax,n,Hangup()
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0$
{CALLERIDNUM})
exten = s,2,ReceiveFAX(${FAXFILE}.tif)
exten = s,3,Hangup()
exten=h,1,System(/usr/local/bin/fax2mail.1.sh --dest-name Sean
--dest-email ${Sean_email} -f ${FAXFILE})
which looks like it works just fine from the cli:
-- DAHDI/2-1 is ringing
-- Redirecting DAHDI/4-1 to fax extension
-- Hungup 'DAHDI/2-1'
== Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on
'DAHDI/4-1'
-- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax
Detected) in new stack
-- Executing [...@incoming-pstn-line:2] Goto(DAHDI/4-1,
incoming-fax,s,1) in new stack
-- Goto (incoming-fax,s,1)
-- Executing [...@incoming-fax:1] Set(DAHDI/4-1,
FAXFILE=/var/spool/asterisk/fax/200901141711-0) in new stack
-- Executing [...@incoming-fax:2] ReceiveFAX(DAHDI/4-1,
/var/spool/asterisk/fax/200901141711-0.tif) in new stack
-- Executing [...@incoming-fax:3] Hangup(DAHDI/4-1, ) in new stack
== Spawn extension (incoming-fax, s, 3) exited non-zero on 'DAHDI/4-1'
-- Executing [...@incoming-fax:1] System(DAHDI/4-1,
/usr/local/bin/fax2mail.1.sh --dest-name Sean --dest-email
seandar...@gmail.com -f /var/spool/asterisk/fax/200901141711-0) in
new stack
-- Hungup 'DAHDI/4-1'
But it doesn't - no email is ever sent. BUT, if I execute the fax2mail
cmd from the terminal (pasting from the cli output) it sends the email:
/usr/local/bin/fax2mail.1.sh --dest-name Sean --dest-email
seandar...@gmail.com -f /var/spool/asterisk/fax/200901141711-0
Am I screwing up the System() command somehow? Is System() screwed up in
1.6.1?
Any clues how to debug this? I did find one relevant thread
http://asteriskforum.ru/viewtopic.php?p=15629 , which is unfortunatley
in Russian. In that thread someone figured out how to turn on DEBUG for
app_fax. How did you do that?
sean
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Re: [asterisk-users] mISDN BRI Asterisk 1.4
Lee Wilson wrote: Also, I guess at this point it doesn't matter for L1, but should I be using Point-To-Point or Point-To-Multipoint? Thanks Yes, you would still need to configure mISDN correctly as well! And AFAIK you will need to use PTMP, as that is what the router would expect... -- Francesco Peeters Thanks for clarifying I've double-checked that it is running ptmp but still no link lights. Anyone got other suggestions? Regards Lee Are you using an ISDN cross cable? I don't know these cards, but most cards are wired as a DTE type device (TE port like a router or phone) and not a DCE type device (NT box). So you might have Tx-Tx and Rx-Tx instead of Rx-Tx and Tx-Rx... ;-) (Note that ISDN cross cables are definately NOT the same as a CAT5E cross cable!) -- Francesco Peeters Ubuntu all the way! 1 laptop, 1 server, 1 desktop at home and several servers in different locations ___ -- 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] 1.6.1-b4: Can't get fax2mail work from System()
OCG Technical Support wrote: Start with your mail log. Any errors visible? How about system log - PAMpermission errors? Thanks for the quick response. maillog shows nothing if it's executed from the System() call. Obviously maillog shows the outgoing if executed from the terminal, Nothing in syslog. asterisk is running as root. sean ___ -- 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 problems
Hi All, I'm running Asterisk 1.4.22.1 on a CentOS 5 machine fitted with a TDM400P. When I upgraded from Asterisk 1.2.12.1, Zap stopped working. Doing zap show channels etc from the Asterisk CLI results in an error saying there's no such command. The machine has Zaptel 1.2.9.1, which I've tried to repair with make clean, make linux26, then make install but to no avail so I thought it's time to bite the bullet and move to DAHDI. After installing DAHDI 2.1.0.3 I get an error saying that the hardware is already in use. I've tried to uninstall Zaptel as follows: - running /etc/init.d/zaptel stop - running /bin/modprobe -r zaptel - running make clean from the zaptel installation directory However, on rebooting Zap still loads, which causes DAHDI to throw an error. I'm currently downloading the latest version of CentOS with a view to rebuilding this machine to use DAHDI from the start if I can't sort this out - but I'd rather fix what I've got if possible. Any ideas? Thanks in advance, -- Geoff ___ -- 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 problems
Have you tried recompiling/installing the new zaptel source before Asterisk? Geoff Lane wrote: Hi All, I'm running Asterisk 1.4.22.1 on a CentOS 5 machine fitted with a TDM400P. When I upgraded from Asterisk 1.2.12.1, Zap stopped working. Doing zap show channels etc from the Asterisk CLI results in an error saying there's no such command. The machine has Zaptel 1.2.9.1, which I've tried to repair with make clean, make linux26, then make install but to no avail so I thought it's time to bite the bullet and move to DAHDI. After installing DAHDI 2.1.0.3 I get an error saying that the hardware is already in use. I've tried to uninstall Zaptel as follows: - running /etc/init.d/zaptel stop - running /bin/modprobe -r zaptel - running make clean from the zaptel installation directory However, on rebooting Zap still loads, which causes DAHDI to throw an error. I'm currently downloading the latest version of CentOS with a view to rebuilding this machine to use DAHDI from the start if I can't sort this out - but I'd rather fix what I've got if possible. Any ideas? Thanks in advance, ___ -- 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 problems
On Wednesday, January 14, 2009, Jose P. Espinal wrote: Have you tried recompiling/installing the new zaptel source before Asterisk? Thanks for the reply. It's the old Zaptel source that was working with Asterisk 1.2.12.1 and so was already compiled and installed prior to upgrading Asterisk. That said, I did try recompiling/installing Zap followed by recompiling/installing Asterisk, but to no avail. -- Geoff ___ -- 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] Beware of DIDX Super Technologies
I think their issue is that they built their business around cheap support in Asain countries which is a hit or miss. I know that when I pointed out an obvious flaw that made them look stupid I got email that I had a $20.00 credit with them. I never mentioned it because I did not think it was realavent. It seems that they try to go around their issues instead of dealing with them. It amazes me that they are still around. - Original Message - From: Alex Balashov abalas...@evaristesys.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 13, 2009 9:25 AM Subject: Re: [asterisk-users] Beware of DIDX Super Technologies I also get the impression that this is a well-beaten horse, as Google queries such as this one tend to reveal: site:lists.digium.com didx Alex Balashov wrote: Never dealt with them personally, but I have been made to understand by everyone else I know who has in no uncertain terms that: - Technical, billing and business-related issues abound. - Support is pretty much a pointless waste of time as you suggest. - Basic features don't work reliably. - Backend provisioning is broken; numbers no longer designated as active continue to be tested by their system. - Billing continues despite cancellation. - A substantial language and timezone barrier is a big impediment to basic communication about routine service issues and basic customer care. #1 complaint about DIDX specifically is that nobody that speaks English coherently or understands SIP issues on a more than passing level can be reached. That's just what I've heard from the acquaintances I have that deal with DIDX and other SuperTec entities; I cannot claim to know independently, so I can't personally endorse their opinions. And the people I am thinking of aren't really involved in Asterisk or on these lists, and so they are not here to speak for themselves. So please take that as a qualification. Super Technologies has a marketing and product presence under a variety of other domain names and brands, all commonly united by Gratuitous Capitalisation of Nouns, abundant error's in basic grammar, punctuation, structure, and usage of English generally. An additionally common feature is bizarre and eccentric whitespace between words and terminating punctuation. (Want to Use Voip ?) www.virtualphoneline.com www.superivr.net www.carrierx.org www.phone2net.com www.super-phone.com www.ip-pabx.com www.call-x.com www.superpbx.net www.superphonedirect.com www.superphonewireless.com www.schoolmanagementsystem.net www.healthmanagementsystem.net www.superfax.com www.groovytel.com ... and doubtless a number of others. Andrew Joakimsen wrote: I assume most people here know what a joke DIDX is -- but in case you didn't already know, please avoid these people. Basic features of their service don't work, their tech support refuses/drags their feet to fix them for a month and if you post publicly about them, they terminate your service. Instead of investing their effort in reading mailinglists to terminate customers maybe they should invest their efforts in fixing the issues with their service first. This is all despite the fact that they don't control the numbers they sell -- that I understood and can deal with (it never was an issue since most of the numbers we had with them were from Global Crossing -- Vendor # 701534 in their system) Hell, I was planning to get off their service anyways, if they would have allowed me time to properly port out the numbers, they would not have created an enemy for life. -- Forwarded message -- From: Rehan Allah Wala re...@supertec.com Date: Sat, Jan 10, 2009 at 13:56 Subject: Your DIDX account To: Andrew Joakimsen joakim...@gmail.com Cc: muneeb @ supertec. com mun...@supertec.com, suza...@supertec.com Thank You for this email Andrew, Please move your numbers in next 3 days somewhere, we will close your account as per your request on Tuesday. Rehan On Sat, Jan 3, 2009 at 13:09, Trixter aka Bret McDanel trix...@0xdecafbad.com wrote: On Sat, 2009-01-03 at 12:14 -0500, Andrew Joakimsen wrote: Can you look at ticket # 702556000194? This is very simple: apparently it isnt. Asterisk is down, I am simulating that with the command stop now, Calls should then go to the failover SIP address, but they do not. I have been back and forth for weeks with your support and they do not figure it out. I am not even sure they understand what I am saying. is this related to the below request for a non-profit doing a telethon? If it is I am confused by it. If it isnt, I am unsure what ticket system you refer to. Additionally I am unsure what your setup is since you havent even provided more information. Odds are the equipment that is supposed to do the failover isnt even asterisk. Further I do not think that its a
Re: [asterisk-users] Block Caller ID
You can try blocking the caller ID in the dial plan. Not sure how that will affect the CDR's. If it does not show up in there in the dial plan you can set a variable to the caller ID then set it to be blank and on hangup update the CDR's. - Original Message - From: Sriram To: asterisk-users@lists.digium.com Sent: Friday, October 10, 2008 6:52 PM Subject: [asterisk-users] Block Caller ID Hi Is there any way to stop Asterisk from sending Caller ID display on the softphones ? I;ve E1 PRIs and SIP extensions , i need to stop caller ID from appearing on the softphones ...but in CDRs caller Ids should show - so please dont suggest to set blockcallerid=yes in zapata.conf ;) Thanks Sriram -- ___ -- 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] Beware of DIDX Super Technologies
More than just support - also core engineering. Dovid Bender wrote: I think their issue is that they built their business around cheap support in Asain countries which is a hit or miss. I know that when I pointed out an obvious flaw that made them look stupid I got email that I had a $20.00 credit with them. I never mentioned it because I did not think it was realavent. It seems that they try to go around their issues instead of dealing with them. It amazes me that they are still around. - Original Message - From: Alex Balashov abalas...@evaristesys.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 13, 2009 9:25 AM Subject: Re: [asterisk-users] Beware of DIDX Super Technologies I also get the impression that this is a well-beaten horse, as Google queries such as this one tend to reveal: site:lists.digium.com didx Alex Balashov wrote: Never dealt with them personally, but I have been made to understand by everyone else I know who has in no uncertain terms that: - Technical, billing and business-related issues abound. - Support is pretty much a pointless waste of time as you suggest. - Basic features don't work reliably. - Backend provisioning is broken; numbers no longer designated as active continue to be tested by their system. - Billing continues despite cancellation. - A substantial language and timezone barrier is a big impediment to basic communication about routine service issues and basic customer care. #1 complaint about DIDX specifically is that nobody that speaks English coherently or understands SIP issues on a more than passing level can be reached. That's just what I've heard from the acquaintances I have that deal with DIDX and other SuperTec entities; I cannot claim to know independently, so I can't personally endorse their opinions. And the people I am thinking of aren't really involved in Asterisk or on these lists, and so they are not here to speak for themselves. So please take that as a qualification. Super Technologies has a marketing and product presence under a variety of other domain names and brands, all commonly united by Gratuitous Capitalisation of Nouns, abundant error's in basic grammar, punctuation, structure, and usage of English generally. An additionally common feature is bizarre and eccentric whitespace between words and terminating punctuation. (Want to Use Voip ?) www.virtualphoneline.com www.superivr.net www.carrierx.org www.phone2net.com www.super-phone.com www.ip-pabx.com www.call-x.com www.superpbx.net www.superphonedirect.com www.superphonewireless.com www.schoolmanagementsystem.net www.healthmanagementsystem.net www.superfax.com www.groovytel.com ... and doubtless a number of others. Andrew Joakimsen wrote: I assume most people here know what a joke DIDX is -- but in case you didn't already know, please avoid these people. Basic features of their service don't work, their tech support refuses/drags their feet to fix them for a month and if you post publicly about them, they terminate your service. Instead of investing their effort in reading mailinglists to terminate customers maybe they should invest their efforts in fixing the issues with their service first. This is all despite the fact that they don't control the numbers they sell -- that I understood and can deal with (it never was an issue since most of the numbers we had with them were from Global Crossing -- Vendor # 701534 in their system) Hell, I was planning to get off their service anyways, if they would have allowed me time to properly port out the numbers, they would not have created an enemy for life. -- Forwarded message -- From: Rehan Allah Wala re...@supertec.com Date: Sat, Jan 10, 2009 at 13:56 Subject: Your DIDX account To: Andrew Joakimsen joakim...@gmail.com Cc: muneeb @ supertec. com mun...@supertec.com, suza...@supertec.com Thank You for this email Andrew, Please move your numbers in next 3 days somewhere, we will close your account as per your request on Tuesday. Rehan On Sat, Jan 3, 2009 at 13:09, Trixter aka Bret McDanel trix...@0xdecafbad.com wrote: On Sat, 2009-01-03 at 12:14 -0500, Andrew Joakimsen wrote: Can you look at ticket # 702556000194? This is very simple: apparently it isnt. Asterisk is down, I am simulating that with the command stop now, Calls should then go to the failover SIP address, but they do not. I have been back and forth for weeks with your support and they do not figure it out. I am not even sure they understand what I am saying. is this related to the below request for a non-profit doing a telethon? If it is I am confused by it. If it isnt, I am unsure what ticket system you refer to. Additionally I am unsure what your setup is since you havent even provided more information. Odds are the equipment that is
Re: [asterisk-users] Zap problems
Zaptel 1.2.9.1 will not work with Asterisk 1.4.22. I would recommend you install Zaptel 1.4.12.1 or go to DAHDI. The first thing you need to do is erase all the zaptel modules from the /lib/modules/kernel version directory and do a depmod -a to make sure only the new DAHDI or Zaptel modules get loaded. If you install DAHDI and do not erase the zaptel modules you will get errors. On Wed, 2009-01-14 at 19:16 -0400, Jose P. Espinal wrote: Have you tried recompiling/installing the new zaptel source before Asterisk? Geoff Lane wrote: Hi All, I'm running Asterisk 1.4.22.1 on a CentOS 5 machine fitted with a TDM400P. When I upgraded from Asterisk 1.2.12.1, Zap stopped working. Doing zap show channels etc from the Asterisk CLI results in an error saying there's no such command. The machine has Zaptel 1.2.9.1, which I've tried to repair with make clean, make linux26, then make install but to no avail so I thought it's time to bite the bullet and move to DAHDI. After installing DAHDI 2.1.0.3 I get an error saying that the hardware is already in use. I've tried to uninstall Zaptel as follows: - running /etc/init.d/zaptel stop - running /bin/modprobe -r zaptel - running make clean from the zaptel installation directory However, on rebooting Zap still loads, which causes DAHDI to throw an error. I'm currently downloading the latest version of CentOS with a view to rebuilding this machine to use DAHDI from the start if I can't sort this out - but I'd rather fix what I've got if possible. Any ideas? Thanks in advance, ___ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] bridge 2 calls
I use post variables. I found this on the web. Forgot where I got it from (sorry that I can't give you credit). ?php //Connect to the Asterisk Manager $socket = fsockopen(127.0.0.1,5038, $errno, $errstr); fputs($socket, Action: Login\r\n); fputs($socket, UserName: username\r\n); fputs($socket, Secret: password\r\n); fputs($socket, Events: off\r\n\r\n); fputs($socket, \r\n\r\n); fputs($socket, Action: Originate\r\n); fputs($socket, Channel: SIP/.$_POST['first_call'].@my_peer\r\n); fputs($socket, Context: mycontext\r\n); fputs($socket, Exten: .$_POST['local_exten'].\r\n); fputs($socket, Priority: 1\r\n); fputs($socket, Callerid: 5551212\r\n); fputs($socket, Timeout: 10\r\n); fputs($socket, Variable: FOO=.$my_var.\r\n); fputs($socket, \r\n\r\n); fputs($socket, \r\n); fputs($socket, Action: Logoff\r\n\r\n); fclose($socket); ? - Original Message - From: Nick Wolf new...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 06, 2009 12:18 PM Subject: Re: [asterisk-users] bridge 2 calls I am also interested in establishing a three way conversation using a simple webpage. I wonder if anyone can provide some help on that. On Tue, Jan 6, 2009 at 7:29 AM, amit mehta amit.magn...@gmail.com wrote: Hi Rilawich, I worked recently on it and that is why can give you the idea how i achived it. You can write an PHP script to get the number and name of the customer.You can phpself to the script.Then you can use an API script to use that number to orignate the call.The channel will be used to call the asterisk internal agent and the other line will call the number that was input by the customer and bridge the call. Hope this might help you. Regards, Amit Mehta Cell: +91 9898340962 On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com wrote: Hi all, I want to build a web page for user to input a phone number. Then, the number will input to asterisk and it will makes call. At that moment, asterisk will make another call to a internal ext. Finally asterisk will bridge 2 calls together for conversion. Does asterisk can do it? How? Thanks, Ango ___ -- 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] Block Caller ID
- Original Message - *From:* Sriram mailto:d_r_sri...@hotmail.com *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Sent:* Friday, October 10, 2008 6:52 PM *Subject:* [asterisk-users] Block Caller ID Hi Is there any way to stop Asterisk from sending Caller ID display on the softphones ? I;ve E1 PRIs and SIP extensions , i need to stop caller ID from appearing on the softphones ...but in CDRs caller Ids should show - so please dont suggest to set blockcallerid=yes in zapata.conf ;) Thanks Sriram ___ -- 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 maybe a better solution is to set the callerid to anonymous or something else and use the cdr userfield to set the callerid. so you still have the information and the client doesnt see the callerid in any way. best regards ___ -- 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] Dropping this SIP message, it's incomplete
I am getting this Error on my Asterisk. How to solve it ? ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete. ___ -- 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 problems
On Wednesday, January 14, 2009, Carlos Chavez wrote: Zaptel 1.2.9.1 will not work with Asterisk 1.4.22. I would recommend you install Zaptel 1.4.12.1 or go to DAHDI. Thanks for the reply. Uninstalling DAHDI and switching to Zap 1.4 did the trick. I can now make calls to and from the PSTN and also use my analogue phone via the TDM card. Once I've sorted out the rest of the config and have time to do things properly I'll do some research into DAHDI with a view to rebuilding the Asterisk box. Hopefully, I'll be able to use most of the configuration I've done to get this far! Thanks again, -- Geoff ___ -- 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] bridge 2 calls
None of these examples actually create a 3-way call, which is, unless I am mistaken, the original request. An incoming/outgoing call gets bridged to a local channel alright, but then how do you bridge that call to yet another call?. I did try some alternatives and the only way I found is by using a meeting room. Not too elegant in my opinion although it works nicely. If anyone knows of a better way please tell me. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender Sent: Wednesday, January 14, 2009 6:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] bridge 2 calls I use post variables. I found this on the web. Forgot where I got it from (sorry that I can't give you credit). ?php //Connect to the Asterisk Manager $socket = fsockopen(127.0.0.1,5038, $errno, $errstr); fputs($socket, Action: Login\r\n); fputs($socket, UserName: username\r\n); fputs($socket, Secret: password\r\n); fputs($socket, Events: off\r\n\r\n); fputs($socket, \r\n\r\n); fputs($socket, Action: Originate\r\n); fputs($socket, Channel: SIP/.$_POST['first_call'].@my_peer\r\n); fputs($socket, Context: mycontext\r\n); fputs($socket, Exten: .$_POST['local_exten'].\r\n); fputs($socket, Priority: 1\r\n); fputs($socket, Callerid: 5551212\r\n); fputs($socket, Timeout: 10\r\n); fputs($socket, Variable: FOO=.$my_var.\r\n); fputs($socket, \r\n\r\n); fputs($socket, \r\n); fputs($socket, Action: Logoff\r\n\r\n); fclose($socket); ? - Original Message - From: Nick Wolf new...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 06, 2009 12:18 PM Subject: Re: [asterisk-users] bridge 2 calls I am also interested in establishing a three way conversation using a simple webpage. I wonder if anyone can provide some help on that. On Tue, Jan 6, 2009 at 7:29 AM, amit mehta amit.magn...@gmail.com wrote: Hi Rilawich, I worked recently on it and that is why can give you the idea how i achived it. You can write an PHP script to get the number and name of the customer.You can phpself to the script.Then you can use an API script to use that number to orignate the call.The channel will be used to call the asterisk internal agent and the other line will call the number that was input by the customer and bridge the call. Hope this might help you. Regards, Amit Mehta Cell: +91 9898340962 On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com wrote: Hi all, I want to build a web page for user to input a phone number. Then, the number will input to asterisk and it will makes call. At that moment, asterisk will make another call to a internal ext. Finally asterisk will bridge 2 calls together for conversion. Does asterisk can do it? How? Thanks, Ango ___ -- 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
[asterisk-users] OT - Differences between modprobe and insmod
hello, Here (http://updates.xorcom.com/astribank/bristuff/1.4/INSTALL.html) you can read : cd qozap modprobe zaptel insmod qozap.o (for kernel 2.4) insmod qozap.ko (for kernel 2.6) ztcfg I thought modprobe was a replacement for insmod. Can someone be kind enough to explain : 1. the difference between modprobe and insmod, 2. why should both commands be issued, 3. how modprobe and insmod compare with statements included in /etc/default/zaptel in Debian systems Regards ___ -- 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] Has anyone used FaxGateway()
Hi, I've been trying to use the FaxGateway application to send T.38 out over Zaptel using asterisk but I don't seem to be having any luck. I'm executing it in the dialplan like: FaxGateway(Zap/g0/[number]) Has anyone had any luck using this thing and can enlighten me on how it's supposed to be used? Thanks. ___ -- 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] Has anyone used FaxGateway()
Well, T.38 works over IP, not TDM... James Lamanna wrote: Hi, I've been trying to use the FaxGateway application to send T.38 out over Zaptel using asterisk but I don't seem to be having any luck. I'm executing it in the dialplan like: FaxGateway(Zap/g0/[number]) Has anyone had any luck using this thing and can enlighten me on how it's supposed to be used? Thanks. ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] call transfer in CDR
Hi, I wonder how I can relate the CDR records for the case of call transfer. I can't find their relationship in CDR. Any can advice? ango ___ -- 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] IAX Java Softphone?
Hi all, thanks Tim and Mexuar for releasing this here... I have already taken the source - and compiled a little java applet which is self signed to test the whole thing. I will put it on my site (and allow users to enter host/user/pass/Calling Number,Calling Name,Number to dial...) for demo usage I would be happy to get some feedback about problems - because i am interessted to integrate it in my callcenter project Tim - can you tell me which audio features it does have - as far as i can see there is alaw and gsm - is there also an echo canceller - jitter buffer ? I will post it here as soon as i have the page up ... regards, Wolfgang Tim Panton schrieb: I'm delighted to be able to say that as part of the agreement on my departure from Mexuar, the Corraleta applet source code Westhawk Ltd wrote for them has been released under the GPL. it is available for download at : http://www.mexuar.com/files/corraleta_sdk.rar Tim. On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ 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 ___ -- 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
