Re: [asterisk-users] Any free video (or audio) softphone VOIP client under Linux with touchscreen friendly interface ?
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 ? http://www.qutecom.org it is successor to openwengo --- Marek Cervenka === ___ -- 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] What are the various models of DID providers
On Wed, Jan 14, 2009 at 7:47 AM, Jai Rangi jpra...@gmail.com wrote: Alex, I must say wow, great explanation. It was a wonderful reading. Thanks to everyone who made this interesting reading! You're all invited to argue about this tomorrow, Friday the 15Th of January at 12 Noon EST on the VoIP Users Conference. http://www.voipusersconference.org IRC #voip-users-conference Specifically, how to join the call: http://www.voipusersconference.org/page/page/list ___ -- 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 16:09:02 +0100, philipp-chemn...@gmx.de wrote: setting the caller ID works perfect. Detecting if a caller is or isn't registered is the problem. I'm using sip. wouldnt ChanIsAvail() or regexten/regcontext settings in sip.conf assist in this ? -- 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] evaluate SIP response codes in dialplan
Joshua Colp schrieb:
- 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
I see. I thought HANGUPCAUSE works only with zaptel. I will give it a try.
thanks
klaus
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.
___
-- 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()
IIRC FaxGateway is intelligent and works in both directions. What are the problems? klaus Alex Balashov schrieb: 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 ___ -- 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
14 jan 2009 kl. 14.02 skrev Klaus Darilion: 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? Check the HANGUPCAUSE, it's much more detailed than DIALSTATUS. We do map the SIP (and all other protocol errors in various channel drivers) codes to ISDN hangup causes, which gives you much more information about why a call failed. The conversion we're using follows the RFC, and where that doesn't cover it, Cisco's documentation. /Olle ___ -- 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
14 jan 2009 kl. 18.57 skrev Philipp Kempgen: 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. Which is incorrect. We don't want to add expose every protocol to the dialplan if not needed. As Josh and I've stated, we have the HANGUPCAUSE that gives you this level of detail, but in a multiprotocol way. The most important feature of Asterisk is that it's a multiprotocol PBX. Even if I think there's only one protocol for the future, there's still a lot of old stuff out there and the beauty is that I can produce services in asterisk covering all of these without knowing the details of all these protocols. It would be really bad if I had to write one app for every protocol covered by my dialplan. /O ___ -- 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] OT - Differences between modprobe and insmod
Just google for your subject. short: insmod just tries to load one module. modprobe checks dependencies and loads needed kernel modules too. klaus Olivier schrieb: 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 ___ -- 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 Thursday, January 15, 2009, D Tucny wrote: It's so much nicer to use packages, in the case of CentOS, RPMs... that way everything installed is owned by the package and removal of the package removes most of what was installed... Thanks for the reply. I must be missing something, since all I've found are the tarballs at asterisk.org and mirrors. However, I'm going to rebuild once I've proved it all works (possibly on a machine of better specification) so it would be good to know where the packages can be found. 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] IAX Java Softphone?
Tim Panton ha scritto: [ ... snip .. ] 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. Thank you very much for the tip, I'll have a look soon. - 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 ;-) Ok! I see ;-)! 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 ___ -- 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] OT - Differences between modprobe and insmod
On Thu, Jan 15, 2009 at 02:13:58AM +0100, Olivier wrote: 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 should also point out that those are left-overs from the old INSTALL page. They are under: === Manual Drivers Loading The following section is added unmodified from the original document. I don't really agree with it, though This is nice for manually testing your installation. The problem with that is that it tends to break on next reboot :-) Older versions of qozap had issues with multiple runs of ztcfg. From what I understnd, those issues have been resolved in recent versions. This also reduces the level of voodoo required and allows easier scripting. I thought modprobe was a replacement for insmod. Can someone be kind enough to explain : 1. the difference between modprobe and insmod, In this specific case I believe modprobe would have worked as well. Not really sure. 2. why should both commands be issued, Because qozap was not yet installed onto the system directory. Normally you'd just run 'modprobe qozap' and it would also pull zaptel with it. 3. how modprobe and insmod compare with statements included in /etc/default/zaptel in Debian systems An alternative for manual testing. You forgot to mention /etc/modules on a Debian system, BTW. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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, here you can find the demo site: http://www.yosd.at/corraleta/ I have also opend a forum for further discussion of the corraleta sdk... http://www.yosd.at/index.php?option=com_joomlaboardItemid=39func=showcatcatid=7 regards, Wolfgang Wolfgang Pichler schrieb: 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 ___ -- 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 15 Jan 2009, at 07:30, Wolfgang Pichler wrote: 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. That was quick :-) 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 don't think the GSM codec is actually in there, from memory it does ULAW/ALaw and Slin There is a jitterbuffer of sorts. I never managed to get the echo canceller to work, although the code for it is in the codebase. I will post it here as soon as i have the page up ... If you plan to do significant work on it, please could you put it on sourceforge so others can chip in ? (That's kinda the point of GPLing it) 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
Re: [asterisk-users] IAX Java Softphone?
Hi, there is no gsm codec - thats correct - i must have seen something else... (is there a gsm - or other - codec implementation available for free use ?) I will test it further - and if it fits my needs - then i will put some work into it... I will put it on sourceforge if you want - but i will also have no problem if you will create it as new project on sourceforge... (i think you would be the better project owner) regards, Wolfgang Tim Panton schrieb: On 15 Jan 2009, at 07:30, Wolfgang Pichler wrote: 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. That was quick :-) 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 don't think the GSM codec is actually in there, from memory it does ULAW/ALaw and Slin There is a jitterbuffer of sorts. I never managed to get the echo canceller to work, although the code for it is in the codebase. I will post it here as soon as i have the page up ... If you plan to do significant work on it, please could you put it on sourceforge so others can chip in ? (That's kinda the point of GPLing it) 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 ___ -- 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 gues understood his email wrong. Seemed to be that he wante to make 2 calls via the web and bridge them. - Original Message - From: C. Savinovich c.savinov...@itntelecom.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, January 15, 2009 2:46 AM Subject: 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 ___ -- 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] call transfer in CDR
On Thu, Jan 15, 2009 at 4:09 AM, Rilawich Ango maillist...@gmail.com wrote: 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 You may want to read this thread. http://lists.digium.com/pipermail/asterisk-users/2008-January/204856.html Regards, Greyman. ___ -- 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] Dropping this SIP message, it's incomplete
On Thu, Jan 15, 2009 at 12:18 AM, David @ULC ucoms2...@gmail.com wrote: 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. If the error message being reported by Asterisk is correct and there is no CSeq header then Asterisk should and is correct to drop the request. The CSeq header is mandatory in all SIP messages and it's not something that a SIP server should try and accomodate. The fix is to determine which device or server is sending the faulty requests and ask the manufacturer to fix it. Regards, Greyman. ___ -- 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 Stealing
Hi All, I'd appreciate some help on how to implement call stealing. That is, where you dial a code to redirect any call on the system to your handset. I'm getting rid of my BRI service and I'm trying to replace the functionality of my existing ISDN2e PBX (Cybergear Gold) with VOIP and Asterisk. On my ISDN PBX, the short-code *46 does this. For example, if I take a call on my living room extension and need to refer to some paperwork, I can go to the study, pick up that extension, dial *46, and the call is transferred to the study where I can continue the call with the paperwork to hand. It also helps if you take a call for someone else if that person can steal the call from your extension. Call parking provides a partial work-around but it's a pain having to remember to park a call before moving location. I haven't found an application for call stealing and can't figure out a way to do this. Can anyone help? TIA, -- 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] IAX Java Softphone?
On 15 Jan 2009, at 10:06, Wolfgang Pichler wrote: Hi, there is no gsm codec - thats correct - i must have seen something else... (is there a gsm - or other - codec implementation available for free use ?) I think there is an LGPL gsm implementation in java. I will test it further - and if it fits my needs - then i will put some work into it... I will put it on sourceforge if you want - but i will also have no problem if you will create it as new project on sourceforge... (i think you would be the better project owner) My friends tell me that googlecode is good too. For personal reasons I'm not keen to be the project owner, but I will contribute when I can. 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
Re: [asterisk-users] evaluate SIP response codes in dialplan
Johansson Olle E schrieb: 14 jan 2009 kl. 18.57 skrev Philipp Kempgen: 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. Which is incorrect. We don't want to add expose every protocol to the dialplan if not needed. As Josh and I've stated, we have the HANGUPCAUSE that gives you this level of detail, but in a multiprotocol way. The most important feature of Asterisk is that it's a multiprotocol PBX. Even if I think there's only one protocol for the future, there's still a lot of old stuff out there and the beauty is that I can produce services in asterisk covering all of these without knowing the details of all these protocols. It would be really bad if I had to write one app for every protocol covered by my dialplan. That's OK. HANGUPCAUSE is OK. Nevertheless a configurable mapping cause codes - SIP response codes would be nice :-) 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] evaluate SIP response codes in dialplan
That is very nice, but where are the HANGUPCAUSE values documented? Thanks. on Thursday 01/15/2009 Johansson Olle E(o...@edvina.net) wrote 14 jan 2009 kl. 14.02 skrev Klaus Darilion: 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? Check the HANGUPCAUSE, it's much more detailed than DIALSTATUS. We do map the SIP (and all other protocol errors in various channel drivers) codes to ISDN hangup causes, which gives you much more information about why a call failed. The conversion we're using follows the RFC, and where that doesn't cover it, Cisco's documentation. /Olle ___ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] Call Stealing
and if you use the trasnfer app whit the features chann? David 2009/1/15 Geoff Lane ge...@gjctech.co.uk Hi All, I'd appreciate some help on how to implement call stealing. That is, where you dial a code to redirect any call on the system to your handset. I'm getting rid of my BRI service and I'm trying to replace the functionality of my existing ISDN2e PBX (Cybergear Gold) with VOIP and Asterisk. On my ISDN PBX, the short-code *46 does this. For example, if I take a call on my living room extension and need to refer to some paperwork, I can go to the study, pick up that extension, dial *46, and the call is transferred to the study where I can continue the call with the paperwork to hand. It also helps if you take a call for someone else if that person can steal the call from your extension. Call parking provides a partial work-around but it's a pain having to remember to park a call before moving location. I haven't found an application for call stealing and can't figure out a way to do this. Can anyone help? TIA, -- 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 -- (\__/) (='.'=)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] evaluate SIP response codes in dialplan
15 jan 2009 kl. 12.42 skrev Klaus Darilion: Johansson Olle E schrieb: 14 jan 2009 kl. 18.57 skrev Philipp Kempgen: 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. Which is incorrect. We don't want to add expose every protocol to the dialplan if not needed. As Josh and I've stated, we have the HANGUPCAUSE that gives you this level of detail, but in a multiprotocol way. The most important feature of Asterisk is that it's a multiprotocol PBX. Even if I think there's only one protocol for the future, there's still a lot of old stuff out there and the beauty is that I can produce services in asterisk covering all of these without knowing the details of all these protocols. It would be really bad if I had to write one app for every protocol covered by my dialplan. That's OK. HANGUPCAUSE is OK. Nevertheless a configurable mapping cause codes - SIP response codes would be nice :-) Absolutely - contact me off line to discuss such a project :-) In the meantime, we could document this a bit better. /O ___ -- 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
15 jan 2009 kl. 13.02 skrev John covici: That is very nice, but where are the HANGUPCAUSE values documented? That's the issue... include/asterisk/causes.h is a good reference for now. /O Thanks. on Thursday 01/15/2009 Johansson Olle E(o...@edvina.net) wrote 14 jan 2009 kl. 14.02 skrev Klaus Darilion: 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? Check the HANGUPCAUSE, it's much more detailed than DIALSTATUS. We do map the SIP (and all other protocol errors in various channel drivers) codes to ISDN hangup causes, which gives you much more information about why a call failed. The conversion we're using follows the RFC, and where that doesn't cover it, Cisco's documentation. /Olle ___ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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 --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 host id
So i made a backup long time ago of the g729 license file for one of my servers, problem is I dont remember which one. Anybody know how I can identify which server this license file belongs to? ___ -- 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?
Dmitry Andrianov wrote: Did I miss something? Is Asterisk capable of handling 16KHz audio already? Can it mix 16KHz streams in the meetme rooms? Can it downsample them to 8kHz for Zap channels? Asterisk 1.6 can handle 16KHz streams and resample between 8KHz and 16KHz. The current conferencing implementation is 8KHz only, but the new one that Josh Colp has been working on will be able to support 16KHz (and higher, presumably) conference mixing. It is scheduled to be in Asterisk 1.6.2, although it will be a parallel implementation, and won't affect app_meetme. Who needs 16khz codec if the rest of the system cannot do anything about it? Very true :-) -- 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
[asterisk-users] R2
hi i am reading about new codecs and new stuff to be added to asterisk. (and i say thanks to all the guys who are working to add all the new features). will be R2 added to the main core of asterisk like ISDN? Thanks David -- (\__/) (='.'=)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] Has anyone used FaxGateway()
Hi,
Here's part of the log that I see.
In this case I'm testing on a box that unfortunately doesn't have a
PRI connection.
I've so far tested with just voice calls so far, but as you can see,
FaxGateway can't even dial out to the SIP trunk properly.
Here's also what the dialplan looks like:
exten = _1NXXNXX,1(faxtest),FaxGateway(SIP/vitel-outbound/${EXTEN},-1)
And the log:
[Jan 14 21:04:04] VERBOSE[8110] logger.c: -- Executing
[1xxx...@from-internal:14] FaxGateway(SIP/xx-098befd0,
SIP/vitel-outbound/1xx|-1) in new stack
[Jan 14 21:04:04] WARNING[8110] rtp.c: Unable to set TOS to 184
[Jan 14 21:04:04] WARNING[8110] udptl.c: UDPTL unable to set TOS to 184
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - transmit entry.
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: FaxGw - transmit -
type SIP destination vitel-outbound/1xx
[Jan 14 21:04:04] VERBOSE[8110] logger.c: Called vitel-outbound/1xx
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - after ast_call.
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - waiting for
activity on channels
[Jan 14 21:04:04] ERROR[7258] chan_sip.c: Got error on T.38 initial
invite. Bailing out.
[Jan 14 21:04:04] DEBUG[7258] chan_sip.c: change_t38_state chnaged state to: 0
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - something
happend on peer
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - AST_FRAME_CONTROL
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - AST_CONTROL_BUSY
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - connections
build - ready 0 and erady to talk - 1
[Jan 14 21:04:04] ERROR[8110] app_faxgateway.c: failed to get
remote_channel SIP vitel-outbound/1xx
[Jan 14 21:04:04] NOTICE[8110] app_faxgateway.c: FaxGateway exit with CONGESTION
[Jan 14 21:04:04] WARNING[8110] app_faxgateway.c: Transmission error
___
-- 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] G729 host id
Jon Weisman wrote: So i made a backup long time ago of the g729 license file for one of my servers, problem is I dont remember which one. Anybody know how I can identify which server this license file belongs to? Use the 'asthostid' tool to get the Host-ID for the candidate servers, and compare to the Host-ID in the license file. -- 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
[asterisk-users] problem with PlayDTMF: no error but no tone
Hi to all i'm using PlayDTMF with AJAM, after the authentication, i make a request like this: host:8088/asterisk/mxml?action=PlayDTMFChannel=SIP/200-sdadsadkioahDigit=1 the result is: ajax-response response type='object' id='unknown'generic response='Success' message='DTMF successfully queued' //response /ajax-response But i can't heard nothing on the channel, i've tried to send the tone both on channel and link, but with no results. If i use normal dtmf from keyboards they works properly. What can i check? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- 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] Call Stealing
Why not use call-conferencing? If you transferred your call into a conference room, you could join the conference from any extension on your *. When the caller hangs up, just end the conference. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane Sent: Thursday, January 15, 2009 5:11 AM To: Asterisk Users Subject: [asterisk-users] Call Stealing Hi All, I'd appreciate some help on how to implement call stealing. That is, where you dial a code to redirect any call on the system to your handset. I'm getting rid of my BRI service and I'm trying to replace the functionality of my existing ISDN2e PBX (Cybergear Gold) with VOIP and Asterisk. On my ISDN PBX, the short-code *46 does this. For example, if I take a call on my living room extension and need to refer to some paperwork, I can go to the study, pick up that extension, dial *46, and the call is transferred to the study where I can continue the call with the paperwork to hand. It also helps if you take a call for someone else if that person can steal the call from your extension. Call parking provides a partial work-around but it's a pain having to remember to park a call before moving location. I haven't found an application for call stealing and can't figure out a way to do this. Can anyone help? TIA, -- 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 ___ -- 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 Cisco 7971G-GE from SCCP to SIP
Ayman, after you BUY the license/firmware, etc, to cisco, I use 7911G with Astterisk, my xml conf file is in the wiki : ) 2009/1/13 Steve Edwards asterisk@sedwards.com On Tue, 13 Jan 2009, Ayman Boules (Live.COM) wrote: It will be great if someone can help me upgrade a Cisco 7971G-GE to SIP. If so, please email me the detailed instructions to do the upgrade. Where's that link to http://letmegogglethatforyou.com?; I will appreciate it much if you have the latest 8.4(2) firmware (file name: cmterm-7970_7971-sip.8-4-2.cop) and email it to me or send me a link to download it... Oh. Of course. Let's all violate cisco's copyright on a public mailing list :) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 -- http://celord.blogspot.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] Dropping this SIP message, it's incomplete
When I use below line sin extension.conf file
[from-ipkall]
exten = 901835,1,NoOp(from-ipkall)
exten = 901835,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
exten = 901835,3,Dial(Local/200 at internal)
I get below CLI :
*Quote:*
login as: root
r...@192.168.0.2's password:
Last login: Wed Jan 14 13:09:59 2009 from 192.168.0.21
Welcome to VICIDIALNOW!!!
-
For access to the VICIDIAL admin and agent web GUI use this URL:
http://192.168.0.2
username: admin
password: vicidialnow
For access to VtigerCRM use this URL:
http://192.168.0.2/vtigercrm
username: admin
password: admin
For professional support, visit http://www.vicidialnow.com or send an
email to: supp...@vicidialnow.com
-
Don't forget to run update_server_ip everytime you change your IP address
[r...@vicidialnow ~]# asterisk -r
Asterisk 1.2.27, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
details.
This is free software, with components licensed under the GNU General
Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'show license' for details.
=
Connected to Asterisk 1.2.27 currently running on vicidialnow (pid = 2642)
Verbosity is at least 21
-- Executing NoOp(SIP/66.54.140.46-091c1d68, from-ipkall) in new stack
-- Executing NoOp(SIP/66.54.140.46-091c1d68, INSPIRED MKTG/2064949182)
in new stack
-- Executing Dial(SIP/66.54.140.46-091c1d68, Local/200 at internal) in
new stack
-- Called 200 at internal
-- Executing AGI(Local/200 at inter...@default-6781,2, agi://
127.0.0.1:4577/call_log) in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial(Local/200 at inter...@default-6781,2, SIP/200 at
inter...@sip||tTor) in new stack
-- Called 200 at inter...@sip
-- Local/200 at inter...@default-6781,1 is ringing
Jan 14 13:29:25 ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq.
Dropping this SIP message, it's incomplete.
== Spawn extension (from-ipkall, 901835, 3) exited non-zero on
'SIP/66.54.140.46-091c1d68'
== Spawn extension (default, 200 at internal, 2) exited non-zero on
'Local/200 at inter...@default-6781,2'
-- Executing DeadAGI(Local/200 at inter...@default-6781,2, agi://
127.0.0.1:4577/call_log) in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI(Local/200 at inter...@default-6781,2, agi://
127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-0-CANCEL--))
in new stack
-- AGI Script
agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-0-CANCEL--)
completed, returning 0
Jan 14 13:29:33 ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq.
Dropping this SIP message, it's incomplete.
vicidialnow*CLI
When I use
*Quote:*
exten = 20620368XX,1,Ringing ; call ringing
exten = 20620368XX,2,Wait(1) ; Wait 1 second for CID delivery from PRI
exten = 20620368XX,3,Answer ; Answer the line
exten =
20620368XX,4,AGI(agi-VDAD_ALL_inbound.agi,CIDLOOKUPRC-LB-SALESLINE-20620368XX-Closer-park--999-1-TESTCAMP)
exten = 20620368XX,5,Hangup
I get engage tone.
Any help ?
On Thu, Jan 15, 2009 at 5:48 AM, David @ULC ucoms2...@gmail.com wrote:
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] R2
Is in the process of being merged. http://bugs.digium.com/view.php?id=12509 http://reviewboard.digium.com/r/40/ http://www.libopenr2.org/ Moisés Silva On Thu, Jan 15, 2009 at 9:44 AM, David fire ddf...@gmail.com wrote: hi i am reading about new codecs and new stuff to be added to asterisk. (and i say thanks to all the guys who are working to add all the new features). will be R2 added to the main core of asterisk like ISDN? Thanks David -- (\__/) (='.'=)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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] G729 host id
awesome thanks! - Original Message - From: Kevin P. Fleming kpflem...@digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 15, 2009 9:48 AM Subject: Re: [asterisk-users] G729 host id Jon Weisman wrote: So i made a backup long time ago of the g729 license file for one of my servers, problem is I dont remember which one. Anybody know how I can identify which server this license file belongs to? Use the 'asthostid' tool to get the Host-ID for the candidate servers, and compare to the Host-ID in the license file. -- 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 ___ -- 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] R2
thanks for the answer. any idea in wich version it will be merged? thanks 2009/1/15 Moises Silva moises.si...@gmail.com Is in the process of being merged. http://bugs.digium.com/view.php?id=12509 http://reviewboard.digium.com/r/40/ http://www.libopenr2.org/ Moisés Silva On Thu, Jan 15, 2009 at 9:44 AM, David fire ddf...@gmail.com wrote: hi i am reading about new codecs and new stuff to be added to asterisk. (and i say thanks to all the guys who are working to add all the new features). will be R2 added to the main core of asterisk like ISDN? Thanks David -- (\__/) (='.'=)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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] Digium TE220 supported protocol
Hi, Our potentiel next phone provider ask me a question i can't answer for sure, maybe someone here knows ? He says that is equipement only support VN4 protocol or more, or ETSI, however i can't find matching terms in the digium documentation or the chan_dahdi/dahdi/system.conf files... Any idea ? 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
Re: [asterisk-users] Call Stealing
On Thursday, January 15, 2009, David fire wrote: and if you use the trasnfer app whit the features chann? Thanks for the suggestion. I'll see if I can find it in the docs. -- 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] Call Stealing
On Thursday, January 15, 2009, Danny Nicholas wrote: Why not use call-conferencing? If you transferred your call into a conference room, you could join the conference from any extension on your *. When the caller hangs up, just end the conference. Thanks for the reply. AIUI, you need to set up the conference before leaving the extension on which you took the call. If so, call parking would probably be better since that leaves the original extension free for further calls. 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] Call Stealing
Geoff Lane wrote: On Thursday, January 15, 2009, Danny Nicholas wrote: Why not use call-conferencing? If you transferred your call into a conference room, you could join the conference from any extension on your *. When the caller hangs up, just end the conference. Thanks for the reply. AIUI, you need to set up the conference before leaving the extension on which you took the call. If so, call parking would probably be better since that leaves the original extension free for further calls. Thanks again, Would SLA (Shared Line Appearance) work for this? Put call on hold, press button beside flashing light on second handset? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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
[asterisk-users] Patton SmartNode 4638 and ISDN2e
Hello Does anyone have any experience with configuring BT (British Telecom) ISDN2e lines to work with Patton SmartNodes? I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e lines - and in turn connected to our internal LAN. I'm having huge issues configuring the SmartNode to successfully see the ISDN channels - and to be honest, I'm lost as to how to then route those calls to Asterisk? The instructions, user guide and website all assume a knowledge of the technology/terminology that I just don't have :-( Has anyone had to do anything similar, and if so would you be able to provide some guidance in plain English? There's a beer in it for the person that can help me get an incoming call from the ISDN2e line to my Asterisk SIP phone first - and another beer for getting a call from my SIP phone, through Asterisk and out over the SmartNode and ISDN lines :-) Thanks in advance Phil ___ -- 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] Call Stealing
On Thursday, January 15, 2009, Drew Gibson wrote: Would SLA (Shared Line Appearance) work for this? Put call on hold, press button beside flashing light on second handset? Thanks for the reply. I don't think it would work with my hardware. I've got two Nortel 355 analog handsets, one plugged into my TDM400P card and the other via an IAXy ATA; two analog cordless handsets connected via Grandstream SIP ATAs; and three USB phones connected via softphones on two PCs and a Mac. Not a proper VOIP handset among them! As you can probably guess, I cobbled my system together from whatever I could get my hands on and niceties like SLA didn't enter my head at the time! However, SLA is functionally almost the same as call parking. In that system, I transfer the call to extension 700 and the parking system tells me the number (usually 701) I need to dial to retrieve the call. I can then hang up the original handset, go the new handset, and dial the given number to connect to the caller. It's a little more convoluted than SLA, but with the same functionality. -- 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] Call Stealing
Quoth Geoff Lane ge...@gjctech.co.uk... AIUI, you need to set up the conference before leaving the extension on which you took the call. Yes you do. You'd need to explicitly send the call to a conference, listen and remember the conference number. FWIW, Call Stealing is a feature I miss from my Argent PBX :-( It was nice to wander off and be able to grab an existing call to my extension from any phone that I picked up. I've not been able to find a way of doing this in Asterisk. -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.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] Call Stealing
Here's a working scenario from my asterisk - I have a static conference 6350 set up with no password. When a call comes in, I transfer it to 6350. I can then access this call from any extension by dialing 6350. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Brown Sent: Thursday, January 15, 2009 12:20 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call Stealing Quoth Geoff Lane ge...@gjctech.co.uk... AIUI, you need to set up the conference before leaving the extension on which you took the call. Yes you do. You'd need to explicitly send the call to a conference, listen and remember the conference number. FWIW, Call Stealing is a feature I miss from my Argent PBX :-( It was nice to wander off and be able to grab an existing call to my extension from any phone that I picked up. I've not been able to find a way of doing this in Asterisk. -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.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] Voicetronix Openswitch 12 + echo problem
Hi all, Anyone has this card installed and configured without echos between SIP and VPB channels ? I have 2 Openswitch cards and i always have echo problems in Analog Lines. If i operated SIP through SIP i have no echos, but if i try to operate SIP through VPB there is alot of echos in line. I played with Input and Output gain in PAP2 settings without luck. Any help would be appreciate ! Thanks in advance, Gleidson Antonio Henriques ___ -- 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] Asterisk - Trixbox
My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.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] Call Stealing
On Thursday, January 15, 2009, Jeff LaCoursiere wrote: Cordless phones? Sorry, couldn't resist :) I've got some but the range isn't good enough to cover my entire house. Besides which it's bad enough playing find the phone when a cordless handset gets eaten by the settee or wanders off to the next room! ;) -- 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] evaluate SIP response codes in dialplan
Hi Olle, Johansson Olle E schrieb: 14 jan 2009 kl. 18.57 skrev Philipp Kempgen: 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. Which is incorrect. We don't want to add expose every protocol to the dialplan if not needed. The if not needed part causes lots of discussions in this case. As Josh and I've stated, we have the HANGUPCAUSE that gives you this level of detail, but in a multiprotocol way. Some (no so) subtle differences get lost. It would be really bad if I had to write one app for every protocol covered by my dialplan. True. But it would be a plus if you *could* do that in order to fine-tune the behavior if you wanted to. I still think we need a SIP_CAUSE channel variable. :-) 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] Call Stealing
On Thu, 15 Jan 2009, Geoff Lane wrote: On Thursday, January 15, 2009, Drew Gibson wrote: [snip] However, SLA is functionally almost the same as call parking. In that system, I transfer the call to extension 700 and the parking system tells me the number (usually 701) I need to dial to retrieve the call. I can then hang up the original handset, go the new handset, and dial the given number to connect to the caller. It's a little more convoluted than SLA, but with the same functionality. Cordless phones? Sorry, couldn't resist :) j ___ -- 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] Call Stealing
On Thu, 15 Jan 2009, Geoff Lane wrote: On Thursday, January 15, 2009, Jeff LaCoursiere wrote: Cordless phones? Sorry, couldn't resist :) I've got some but the range isn't good enough to cover my entire house. Besides which it's bad enough playing find the phone when a cordless handset gets eaten by the settee or wanders off to the next room! ;) I'm a bit confused as to how your old system exactly worked. When you initially answer the phone (on presumably the wrong extension), what did you do with that handset before getting up and going to the right extension to steal it? Did you just leave it off hook? I'm assuming you had to dial something to park the call before just hanging up the orginal extension. In that case, call parking is really what you are looking for, and you could simulate your old feature by mapping whatever you used to dial to park the call. Then map your old call steal code to retrieve it. If you actually left the first one off hook in the past and walked to the second extension and then stole the call, then perhaps you could map your code to do a call transfer given the channel id (as someone else suggested I think). j ___ -- 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] Warning in CLI: Inringing for peer [PEER] 0
I get this warning in the Asterisk CLI once in a while, and it usually corresponds with a phone not ringing when it should. Warning in CLI: Inringing for peer [PEER] 0 What does it mean and what is the likely cause of this? ___ -- 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
Stefan Schmidt s...@sil.at writes: 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. Adaptive CDR (and custom CDR, if you prefer files) supports user-defined fields. It is very helpful, because you can put in something like CDR(anumber) and CDR(bnumber) and format them correctly, no matter what strange format your SIP peers want their calls in. /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] Call Stealing
On Thursday, January 15, 2009, Jeff LaCoursiere wrote: I'm a bit confused as to how your old system exactly worked. When you initially answer the phone (on presumably the wrong extension), what did you do with that handset before getting up and going to the right extension to steal it? Did you just leave it off hook? I'm assuming you had to dial something to park the call before just hanging up the orginal extension. In that case, call parking is really what you are looking for, and you could simulate your old feature by mapping whatever you used to dial to park the call. Then map your old call steal code to retrieve it. You just leave the phone off the hook, walk to the handset to which you want to transfer the call, then dial the call-steal code. This steals (captures) any active call within the same ring group. You don't need to park the call first. If you actually left the first one off hook in the past and walked to the second extension and then stole the call, then perhaps you could map your code to do a call transfer given the channel id (as someone else suggested I think). AIUI, the syntax for the Transfer() function is Transfer(exten) where exten is the destination extension. AFAICT, it implicitly transfers from the current extension in the dialplan, so you can use it to push a call to another extension but I can't see how to use it pull a call from another extension. Am I missing something, or is there another application that can pull a call? TIA, -- 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] Call Stealing
I'm confused as to why you think leaving a phone off the hook is better than parking the call and hanging up the phone. The phone that's off the hook can't receive any more calls after you've 'pulled' the one it was on the line with, assuming you don't walk back to that phone and subsequently hang it up, making the originating extension effectively useless. Call parking and hanging up the originating extension is actually a more elegant solution in my opinion. --Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane Sent: Thursday, January 15, 2009 3:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Stealing You just leave the phone off the hook, walk to the handset to which you want to transfer the call, then dial the call-steal code. This steals (captures) any active call within the same ring group. You don't need to park the call first. ___ -- 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] Call Stealing
What about Chanspy()? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane Sent: Thursday, January 15, 2009 2:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Stealing On Thursday, January 15, 2009, Jeff LaCoursiere wrote: I'm a bit confused as to how your old system exactly worked. When you initially answer the phone (on presumably the wrong extension), what did you do with that handset before getting up and going to the right extension to steal it? Did you just leave it off hook? I'm assuming you had to dial something to park the call before just hanging up the orginal extension. In that case, call parking is really what you are looking for, and you could simulate your old feature by mapping whatever you used to dial to park the call. Then map your old call steal code to retrieve it. You just leave the phone off the hook, walk to the handset to which you want to transfer the call, then dial the call-steal code. This steals (captures) any active call within the same ring group. You don't need to park the call first. If you actually left the first one off hook in the past and walked to the second extension and then stole the call, then perhaps you could map your code to do a call transfer given the channel id (as someone else suggested I think). AIUI, the syntax for the Transfer() function is Transfer(exten) where exten is the destination extension. AFAICT, it implicitly transfers from the current extension in the dialplan, so you can use it to push a call to another extension but I can't see how to use it pull a call from another extension. Am I missing something, or is there another application that can pull a call? TIA, -- 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 ___ -- 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] how to debug mime-construct with fax2mail?
I'm trying to capture faxes on 1.6.1-beta4. AFAICT, app_fax is working
OK. I'm then using fax2mail to send the fax. That wasn't working, so i
posted for help using the System() cmd, since fax2mail did work from the
command line. But now I realize it's fax2mail and mime-construct itself.
I set up a fax-test context:
[fax-test]
exten=666,1,NoOp( fax-test )
exten=666,2,System(/bin/echo this is a system
test${STRFTIME(${EPOCH},,%H%M)} /opt/system-test)
exten=666,3,System(/usr/local/bin/fax2mail.1.sh --dest-name Sean
--dest-email seandar...@gmail -f /var/spool/asterisk/fax/FAXFILE)
exten=666,n,Hangup
This works fine on the cli. And /opt/system-test captures the /bin/echo
string.
AND, the fax2mail log - /var/log/asterisk/faxlog - shows that fax2mail
was`called, and there are no errors. So it's not the System() cmd. But
the email is NOT sent.
faxlog:
fax2mail v2.3
Triggered on Thursday, January 15 2009, at 02:45 PM
Called with --dest-name Sean --dest-email seandar...@gmail -f
/var/spool/asterisk/fax/FAXFILE
CallerID number of fax sender =
CallerID name of fax sender = Someone Unknown
Fax number called = 213 666 9505
Destination name = Sean
Destination email address = seandar...@gmail
Fax file name (without .tif extension) = /var/spool/asterisk/fax/FAXFILE
Attachment format conversion = pdf
Set CallerID number of fax sender to unknown number
Fax file /var/spool/asterisk/fax/FAXFILE.tif found.
Converted /var/spool/asterisk/fax/FAXFILE.tif to
/var/spool/asterisk/fax/FAXFILE.pdf.
E-mailed file to seandar...@gmail
Removing destination file /var/spool/asterisk/fax/FAXFILE.pdf
I can run the exact same cmd from the terminal, and it works. The email
is sent. And the fax2mail log looks the same.
asterisk is running as root, I run the command at the terminal as root.
So I getting to think it's somehow mime-construct, which doesn't seem to
have some nice log around, even if run with --debug.
Any help really appreciated. I'm puzzled as hell.
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
Re: [asterisk-users] how to debug mime-construct with fax2mail?
Have you tried your system stuff under su - asterisk? Once it works that
way, the system() command will work.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Thursday, January 15, 2009 2:45 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to debug mime-construct with fax2mail?
I'm trying to capture faxes on 1.6.1-beta4. AFAICT, app_fax is working
OK. I'm then using fax2mail to send the fax. That wasn't working, so i
posted for help using the System() cmd, since fax2mail did work from the
command line. But now I realize it's fax2mail and mime-construct itself.
I set up a fax-test context:
[fax-test]
exten=666,1,NoOp( fax-test )
exten=666,2,System(/bin/echo this is a system
test${STRFTIME(${EPOCH},,%H%M)} /opt/system-test)
exten=666,3,System(/usr/local/bin/fax2mail.1.sh --dest-name Sean
--dest-email seandar...@gmail -f /var/spool/asterisk/fax/FAXFILE)
exten=666,n,Hangup
This works fine on the cli. And /opt/system-test captures the /bin/echo
string.
AND, the fax2mail log - /var/log/asterisk/faxlog - shows that fax2mail
was`called, and there are no errors. So it's not the System() cmd. But
the email is NOT sent.
faxlog:
fax2mail v2.3
Triggered on Thursday, January 15 2009, at 02:45 PM
Called with --dest-name Sean --dest-email seandar...@gmail -f
/var/spool/asterisk/fax/FAXFILE
CallerID number of fax sender =
CallerID name of fax sender = Someone Unknown
Fax number called = 213 666 9505
Destination name = Sean
Destination email address = seandar...@gmail
Fax file name (without .tif extension) = /var/spool/asterisk/fax/FAXFILE
Attachment format conversion = pdf
Set CallerID number of fax sender to unknown number
Fax file /var/spool/asterisk/fax/FAXFILE.tif found.
Converted /var/spool/asterisk/fax/FAXFILE.tif to
/var/spool/asterisk/fax/FAXFILE.pdf.
E-mailed file to seandar...@gmail
Removing destination file /var/spool/asterisk/fax/FAXFILE.pdf
I can run the exact same cmd from the terminal, and it works. The email
is sent. And the fax2mail log looks the same.
asterisk is running as root, I run the command at the terminal as root.
So I getting to think it's somehow mime-construct, which doesn't seem to
have some nice log around, even if run with --debug.
Any help really appreciated. I'm puzzled as hell.
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
___
-- 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] Call Stealing
On Thursday, January 15, 2009, David Gibbons wrote: I'm confused as to why you think leaving a phone off the hook is better than parking the call and hanging up the phone. Simply that you don't have to remember to park the call. With call parking, if you forget to park the call before moving location you have to return to the original location, park the call, then try again. With call stealing, you can't forget to park the call because it's not required. The phone that's off the hook can't receive any more calls after you've 'pulled' the one it was on the line with, assuming you don't walk back to that phone and subsequently hang it up, making the originating extension effectively useless. The first person to walk by the handset who hears the engaged tone hangs up. I've been using this system for about eight years and it's never been an issue. Also, the originating extension isn't effectively useless because if you hear the phone in the next room ring, you can hang up the originating extension and within a couple of seconds it also rings so that you can take the call. Call parking and hanging up the originating extension is actually a more elegant solution in my opinion. I can see pros and cons to each. However, I (and all my family) are used to call stealing so it would be better if we could duplicate that rather than having to retrain everyone to use call parking. -- 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] how to debug mime-construct with fax2mail?
Have you tried your system stuff under su - asterisk? Once it works that way, the system() command will work. asterisk is running as root, I run the command at the terminal as root. I am guessing he doesn't even have an asterisk user. ___ -- 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] Call Stealing
On Thursday, January 15, 2009, Danny Nicholas wrote: What about Chanspy()? Thanks for the reply, but I suspect it won't do what I want. AIUI, ChanSpy() doesn't transfer the call - it just lets another extension listen in (and join in the conversation in whisper mode). So (AFAICT) the call will be lost if someone hangs up the originating extension. 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] how to debug mime-construct with fax2mail?
Joseph L. Casale wrote: Have you tried your system stuff under su - asterisk? Once it works that way, the system() command will work. asterisk is running as root, I run the command at the terminal as root. I am guessing he doesn't even have an asterisk user. Well I do have an asterisk user, and once spent a weekend trying to run asterisk as asterisk user. But I don't see what this has to do with my problem. The System() cmd works: I can see the log from fax2mail showing it was called, and called with the arguments I expected. So System() did it's thing. What I can't figure what is why fax2mail really works from the command line, but fails to effectively call mime-construct when called from System(). I was hoping someone who has used mime-construct could show me how to debug it. It may be a permissions problem, but since both run as root it seems unlikely. In any event, being able to debug mime-construct would allow me to figure it out. 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
Re: [asterisk-users] how to debug mime-construct with fax2mail?
If you are running the script within Asterisk as root, then it's a path environment issue. My guess(and I run into this with cron jobs all the time) is that the path is different from the command line than the environment that the script runs under. There are times where the fix is to use the fully qualified path when calling stuff and not assume it's in the path. Lyle sean darcy wrote: Joseph L. Casale wrote: Have you tried your system stuff under su - asterisk? Once it works that way, the system() command will work. asterisk is running as root, I run the command at the terminal as root. I am guessing he doesn't even have an asterisk user. Well I do have an asterisk user, and once spent a weekend trying to run asterisk as asterisk user. But I don't see what this has to do with my problem. The System() cmd works: I can see the log from fax2mail showing it was called, and called with the arguments I expected. So System() did it's thing. What I can't figure what is why fax2mail really works from the command line, but fails to effectively call mime-construct when called from System(). I was hoping someone who has used mime-construct could show me how to debug it. It may be a permissions problem, but since both run as root it seems unlikely. In any event, being able to debug mime-construct would allow me to figure it out. 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 ___ -- 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] Call Stealing
hey it is preatty easy now i understand the problem is simple hangup in new location dial steal code for asterisk is just an extension and it should start an AGI the system search for the call in the same group bridge the channel to the current channel asterisk 1.6 or the system search for the call in the same group (AGI) send the channel to a conference (AGI search for the first free conference) join the current channel to the conference (AGI or AGI set a variable whit the conference number) 2009/1/15 Geoff Lane ge...@gjctech.co.uk On Thursday, January 15, 2009, Danny Nicholas wrote: What about Chanspy()? Thanks for the reply, but I suspect it won't do what I want. AIUI, ChanSpy() doesn't transfer the call - it just lets another extension listen in (and join in the conversation in whisper mode). So (AFAICT) the call will be lost if someone hangs up the originating extension. 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 -- (\__/) (='.'=)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] Call Stealing
Look int the ChannelRedirect command. Geoff Lane wrote: Hi All, I'd appreciate some help on how to implement call stealing. That is, where you dial a code to redirect any call on the system to your handset. I'm getting rid of my BRI service and I'm trying to replace the functionality of my existing ISDN2e PBX (Cybergear Gold) with VOIP and Asterisk. On my ISDN PBX, the short-code *46 does this. For example, if I take a call on my living room extension and need to refer to some paperwork, I can go to the study, pick up that extension, dial *46, and the call is transferred to the study where I can continue the call with the paperwork to hand. It also helps if you take a call for someone else if that person can steal the call from your extension. Call parking provides a partial work-around but it's a pain having to remember to park a call before moving location. I haven't found an application for call stealing and can't figure out a way to do this. Can anyone help? TIA, ___ -- 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] R2
That's Digium's folks decision. It was said they wanted it for 1.6.3, but, that's not for sure, as I said, they will decide. On Thu, Jan 15, 2009 at 11:54 AM, David fire ddf...@gmail.com wrote: thanks for the answer. any idea in wich version it will be merged? thanks 2009/1/15 Moises Silva moises.si...@gmail.com Is in the process of being merged. http://bugs.digium.com/view.php?id=12509 http://reviewboard.digium.com/r/40/ http://www.libopenr2.org/ Moisés Silva On Thu, Jan 15, 2009 at 9:44 AM, David fire ddf...@gmail.com wrote: hi i am reading about new codecs and new stuff to be added to asterisk. (and i say thanks to all the guys who are working to add all the new features). will be R2 added to the main core of asterisk like ISDN? Thanks David -- (\__/) (='.'=)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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] Broadcast Phone system (for radio)
this link: http://www.telos-systems.com/techtalk/digiphones/digiphones_4.htm States the following: Generic PBXs will not do for our broadcast application – they just don’t have the features necessary. For example, while lines may certainly be shared to multiple phones, there is no way to switch groups of lines from studio to studio. There is also no way to connect computers for call-screening applications. On the audio side, there is no adaptive hybrid or professional audio outputs. Usually, there is only one or two “Music on Hold” inputs for the entire unit, while we need one for each studio. While you could use a PBX to derive analog lines for the studio telephone interface gear, it will be far superior to make a direct all-digital link. So we will need something like a PBX, but specialized for broadcast. Our company owns 2 radio stations, and they are looking at a new on-air phone system. At the same time, we are looking at installing an Asterisk system for their office PBX. Does anyone know of an asterisk based solution for this type of application? I'm pretty certain Asterisk could handle all the special requirements that this article is claiming a Generic PBX can't 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
[asterisk-users] How to transfer a call from one Asterisk Server to another
Can anyone tell me how I can completely move an established call off of one Asterisk server to another? In our case we have a server with our IVR. Depending upon digits entered, the call can be transferred to any of our other servers depending where the extension or queue reside. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 Our provider which delivers inbound calls to us uses a Sonus gateway. So far, testing has shown that if we transfer the inbound call prior to any media playback, it works. But, if the IVR plays media, then it is failing, with a 500 internal server error being returned. Thanks for any help ___ -- 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] How to transfer a call from one Asterisk Server to another
Are you planning on connecting your two Asterisk servers with SIP or IAX? Check out this tutorial if using SIP: http://hostseries.com/connecting-to-asterisk-servers-via-sip/ You should be able to adapt it to your needs. Good luck! Paul wrote: Can anyone tell me how I can completely move an established call off of one Asterisk server to another? In our case we have a server with our IVR. Depending upon digits entered, the call can be transferred to any of our other servers depending where the extension or queue reside. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 Our provider which delivers inbound calls to us uses a Sonus gateway. So far, testing has shown that if we transfer the inbound call prior to any media playback, it works. But, if the IVR plays media, then it is failing, with a 500 internal server error being returned. Thanks for any help ___ -- 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] Call Stealing
On Thursday 15 January 2009 13:02:32 Geoff Lane wrote: On Thursday, January 15, 2009, Jeff LaCoursiere wrote: Cordless phones? Sorry, couldn't resist :) I've got some but the range isn't good enough to cover my entire house. Besides which it's bad enough playing find the phone when a cordless handset gets eaten by the settee or wanders off to the next room! ;) You could just use the Pickup application: Pickup(ext[@context]) So if extension 101 in context 'incoming' is ringing: Pickup(1...@incoming) -- Tilghman ___ -- 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] Call Stealing
On Thu, 15 Jan 2009, Tilghman Lesher wrote: On Thursday 15 January 2009 13:02:32 Geoff Lane wrote: On Thursday, January 15, 2009, Jeff LaCoursiere wrote: Cordless phones? Sorry, couldn't resist :) I've got some but the range isn't good enough to cover my entire house. Besides which it's bad enough playing find the phone when a cordless handset gets eaten by the settee or wanders off to the next room! ;) You could just use the Pickup application: Pickup(ext[@context]) So if extension 101 in context 'incoming' is ringing: Pickup(1...@incoming) That doesn't work once the call is actually answered by the first extension, though, correct? j -- Tilghman ___ -- 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] multiple registration to sip trunking provider.
a strange problem of multiple sip registrations and peer selection in sip.conf is calling for your suggestions!! let's examine this scenario: some numbers and passwords hidden with HHHs to protect the guilty :) I have 3 distinct sip subscriptions with cordiaip.net provider in US. For each of these i insert in sip.conf (with peer name differences and relefant number/password differences, of course] --- register = 1646H25:hh...@soft1.ny.cordiaip.net/1646H25 [cordiaus1] type=friend secret=H username=1646H25 fromuser=1646H25 fromdomain=soft1.ny.cordiaip.net host=soft1.ny.cordiaip.net call-limit=5 outboundproxy=soft1.ny.cordiaip.net disallow=all allow=gsm allow=alaw allow=ulaw context=DID_cordia insecure=port --- the sip registrations are OK and all seeems fine, BUT i have difficulties to map the incoming call because * is making mistakes in matching the incoming sip INVITE to the relevant peer. Please note that ALL the peers share the very same host and sip port. When i make a call to one of the subscribed cordia number, in sip debug i get a packes similar to this: --- SIP read from 38.98.115.34:5060 --- INVITE sip:16462487...@87.241.44.202 SIP/2.0 Via: SIP/2.0/UDP 38.98.115.34:5060;branch=z9hG4bK22981681-bdb335 To: sip:1646...@38.98.115.34 From: sip:39347...@38.98.115.34;tag=2298168-fdb335 Call-ID: 4926-0-1232058...@38.98.115.7 CSeq: 1 INVITE Contact: sip:393477135...@38.98.115.34:5060;transport=udp Server: Sansay-SIP/8.0 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 201 v=0 o=Sansay-SPX 11 11 IN IP4 38.98.115.9 s=Session Controller c=IN IP4 38.98.115.9 t=0 0 m=audio 15986 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 please note the From: and To: lines, I receive a From: with the caller CID (my mobile phone, in this case) and a To: with the sip number the call is directed to; this seems OK to me. I have read the chan_sip.c source file and it seems that when * receives this invite, it wals through the list of sip users/peers/friends to search for the correct entry from which cloning the sip parameters for the channel (as moh class, call limits, codecs and such) using the host IP as the key (if type=peer) or the caller number (if type=user) getting the values from the From: header. This seems very strange, because the user part of the From: header is potentially ANY number and the host part (and not the port, because is is always 5060 and there is insecure=port in place) in this scenario is not unique due to the 3 peers definitions. Please keep in mind that if i utilize only one registration i have absolutely no problems and can configure * correctly. The problem presents itself ONLY with multiple peers with multiple registrations to the same host/port. I cannot request cordia to forward me the numbers via an unique sip registration (sip trunking) because it seems that they don't offer this service. (but it may well be that i hadn't asked the right question) Can anyone suggest how to implement a correct sip trunking for this scenario, in which I have the incoming calls of the three registration going in a specific context (not the default, see context=DID_cordia in the peer definitions) and the outgoing calls going out via a specific user (so i can choose at the dialplan level with which number i am presenting myself in outgoing calls) I have spent some days trying various combinations of peers and users definitions, going in all cases to crash on the wall of the algorithm * uses to select the correct peer for the incoming calls. ___ -- 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] how to debug mime-construct with fax2mail?
Lyle Giese wrote: If you are running the script within Asterisk as root, then it's a path environment issue. My guess(and I run into this with cron jobs all the time) is that the path is different from the command line than the environment that the script runs under. There are times where the fix is to use the fully qualified path when calling stuff and not assume it's in the path. Lyle You are the man. If we ever meet I owe you a beer, at least one. In the fax2mail script, it just calls mime-construct without a full path. mime-construct on my box is in /usr/local/bin which must not be in the path of the environment System calls are run in. Putting in the fully qualified path made it work. Thanks again. 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
Re: [asterisk-users] Call Stealing
On Thursday 15 January 2009 17:36:31 Jeff LaCoursiere wrote: On Thu, 15 Jan 2009, Tilghman Lesher wrote: On Thursday 15 January 2009 13:02:32 Geoff Lane wrote: On Thursday, January 15, 2009, Jeff LaCoursiere wrote: Cordless phones? Sorry, couldn't resist :) I've got some but the range isn't good enough to cover my entire house. Besides which it's bad enough playing find the phone when a cordless handset gets eaten by the settee or wanders off to the next room! ;) You could just use the Pickup application: Pickup(ext[@context]) So if extension 101 in context 'incoming' is ringing: Pickup(1...@incoming) That doesn't work once the call is actually answered by the first extension, though, correct? That is correct. However, in the original usage scenario, you suggested that you merely be able to pickup a remote ringing extension. This, it will do. -- Tilghman ___ -- 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] how to debug mime-construct with fax2mail?
If you want to email me your fixed script I'll put it up on the web site... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: January 15, 2009 7:08 PM To: Asterisk Users List Subject: Re: [asterisk-users] how to debug mime-construct with fax2mail? Lyle Giese wrote: If you are running the script within Asterisk as root, then it's a path environment issue. My guess(and I run into this with cron jobs all the time) is that the path is different from the command line than the environment that the script runs under. There are times where the fix is to use the fully qualified path when calling stuff and not assume it's in the path. Lyle You are the man. If we ever meet I owe you a beer, at least one. In the fax2mail script, it just calls mime-construct without a full path. mime-construct on my box is in /usr/local/bin which must not be in the path of the environment System calls are run in. Putting in the fully qualified path made it work. Thanks again. 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 ___ -- 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] gtalk and jingle again...
Hello everyone! I just installed the latest asterisk from svn. Now I'm retrying my luck with gtalk and jingle. I have moved so my basic setup has changed a bit... I'm not sure if it helps or hurts. I tried this: call myself: channel originate gtalk/gtalk_account/julienco...@googlemail.com application \ Jack i(system:playback_1)o(system:capture_1) I got some notes about a lot of traffic going on, but no call. Not sure if this is the old jack trouble biting back or something else. I would have tried with a call transfer to isdn, but I don't have real ISDN now. I'll have to check that seperately. If someone with working jingle or gtalk s still up, I'd be happy for a short test call. Usually I was able to receive calls without trouble. Here are my gtalk and jingle accounts. Just drop me a line so I can take you up in my configs. julienco...@googlemail.com julienco...@jabber.org Thanks in any case! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN and routers...
Hello! Sorry for not being able to phrase the problem in one line. My phone situation is this: The calls go over analog line (or NGN/vip) I don't really get to see it. I have got a router with a lot of jacks. One or two of them are for ISDN phones or other ISDN capable devices. Can I use chan_misdn and my good old ISDN card with this setup. Or do I have to get a card, that can handle analog lines. The telephone now connected to the router is analog and quite old. the router is by Samsung, but I couldn't find out, which device it is exactly. will have to wait till I get braille-support for the stupid win-notebook. Too much javascript in the webinterface... :-( Thanks for any good hints on this! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CRTC and FCC Feeds
I don't understand why so many government sites fail to provide some sort of feed to their daily bulletins. What I am venting about in specific are the Canadian CRTC and FCC sites, every day I have to go to the website and when I reach the content, usually it isn't even HTML but a Word or PDF file. So finally today I decided to do something about it and wrote a little app that scrapes their daily releases and displays the information in blogger so I can just add the feeds to my reader. The CRTC site specially dissapoints me because someone who developed the site had the common sense of using Dublin Core in the meta tags or are using a CMS which obviously makes it easy to make available machine readable content. I wrote this system using Python, Beautiful Soup and Google's App. Engine, I will allow comments on both and if there is demand will switch to a PLIGG instance instead of blogger. As a legal note, I make absolutely no claims or warranties that this application actually works or the following blogs display accurate data from the CRTC or FCC's site. Here they are: for the CRTC: http://crtc.gulfpearl.com for the FCC: http://fcc.gulfpearl.com --- Shidan Gouran shidan.gulfpearl.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
[asterisk-users] Portech MV-378 with Asterisk
Has anyone been able to configure portech's mv-378 gateway with asterisk? I did the configuration as per the manual but it does not work. My server sees the portech gateway, but when the gateway is trying to register to my server it fails. It says peer is not suppose to register. The gateway and the asterisk box are on two different location (two network, 2 differrent IP address). I would appreciate any kind of tutorial or advice on how to make it work. 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] CRTC and FCC Feeds
What about this? http://www.thefederalregister.com/rss/department/FEDERAL_COMMUNICATIONS_COMMISSION/ Shidan wrote: I don't understand why so many government sites fail to provide some sort of feed to their daily bulletins. What I am venting about in specific are the Canadian CRTC and FCC sites, every day I have to go to the website and when I reach the content, usually it isn't even HTML but a Word or PDF file. So finally today I decided to do something about it and wrote a little app that scrapes their daily releases and displays the information in blogger so I can just add the feeds to my reader. The CRTC site specially dissapoints me because someone who developed the site had the common sense of using Dublin Core in the meta tags or are using a CMS which obviously makes it easy to make available machine readable content. I wrote this system using Python, Beautiful Soup and Google's App. Engine, I will allow comments on both and if there is demand will switch to a PLIGG instance instead of blogger. As a legal note, I make absolutely no claims or warranties that this application actually works or the following blogs display accurate data from the CRTC or FCC's site. Here they are: for the CRTC: http://crtc.gulfpearl.com for the FCC: http://fcc.gulfpearl.com --- Shidan Gouran shidan.gulfpearl.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 -- 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] Asterisk Upgrade
I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___ -- 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] Broadcast Phone system (for radio)
Ah, But Asterisk if not your Generic PBX! You could do a few things. For each show, (I take it that this is talk radio) You can set up a queue() for each air studio. Callers would then be greeted with a custom greeting that would be unique for each air studio. How you interface with your console (sound board, not phone) would be up to you, you could either have the calls go into a phone patch our you could even use a PC with a Softphone and take the Input and output of its sound card and interface it into your board. (Ground loops and interference notwithstanding) As far as 'Hold music', you have several options: 1 You can use sound files (mp3, gsm, wav, etc.) and have that as your hold music either global (one message for all stations, and admin offices) or you could always get a sound card and 'feed' each air studios program into the queue for the respective air studio call in queue (sound cards have two channels, telephony until now is mono). I don't know much about your current setup, so I was pretty general and conservative on my suggestions but it is defiantly doable and I have done it before for a corporate call in show. Work very well and quality was excellent from caller to broadcast. If you have any questions you call reach me directly. Alex Lopez -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Bob Pierce Sent: Thursday, January 15, 2009 5:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Broadcast Phone system (for radio) this link: http://www.telos-systems.com/techtalk/digiphones/digiphones_4.htm States the following: Generic PBXs will not do for our broadcast application - they just don't have the features necessary. For example, while lines may certainly be shared to multiple phones, there is no way to switch groups of lines from studio to studio. There is also no way to connect computers for call-screening applications. On the audio side, there is no adaptive hybrid or professional audio outputs. Usually, there is only one or two Music on Hold inputs for the entire unit, while we need one for each studio. While you could use a PBX to derive analog lines for the studio telephone interface gear, it will be far superior to make a direct all-digital link. So we will need something like a PBX, but specialized for broadcast. Our company owns 2 radio stations, and they are looking at a new on-air phone system. At the same time, we are looking at installing an Asterisk system for their office PBX. Does anyone know of an asterisk based solution for this type of application? I'm pretty certain Asterisk could handle all the special requirements that this article is claiming a Generic PBX can't 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 ___ -- 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] R2
thanks moises and Digium's folks put it asap please not until 1.6.3 thanks 2009/1/15 Moises Silva moises.si...@gmail.com That's Digium's folks decision. It was said they wanted it for 1.6.3, but, that's not for sure, as I said, they will decide. On Thu, Jan 15, 2009 at 11:54 AM, David fire ddf...@gmail.com wrote: thanks for the answer. any idea in wich version it will be merged? thanks 2009/1/15 Moises Silva moises.si...@gmail.com Is in the process of being merged. http://bugs.digium.com/view.php?id=12509 http://reviewboard.digium.com/r/40/ http://www.libopenr2.org/ Moisés Silva On Thu, Jan 15, 2009 at 9:44 AM, David fire ddf...@gmail.com wrote: hi i am reading about new codecs and new stuff to be added to asterisk. (and i say thanks to all the guys who are working to add all the new features). will be R2 added to the main core of asterisk like ISDN? Thanks David -- (\__/) (='.'=)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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] bridge 2 calls
Thanks all. I think click to call can fulfill my purpose. On Thu, Jan 15, 2009 at 6:10 PM, Dovid Bender asteriskus...@dovid.net wrote: I gues understood his email wrong. Seemed to be that he wante to make 2 calls via the web and bridge them. - Original Message - From: C. Savinovich c.savinov...@itntelecom.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, January 15, 2009 2:46 AM Subject: 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 ___ -- 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] how to debug mime-construct with fax2mail?
OCG Technical Support wrote: If you want to email me your fixed script I'll put it up on the web site... Well I'd be pleased to have any script of mine put up on any web site, but the only thing I did was to hard wire my location of mime-construct: MimeC=/usr/local/bin/mime-construct and the changed all the calls to mime-construct to MimeC. Not very portable :( I suppose what should happen is a test if mime-construct is in the path, and then a search. But this is waay beyond my scripting prowess. 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
Re: [asterisk-users] Digium TE220 supported protocol
On Thu, Jan 15, 2009 at 06:11:59PM +0100, Benoit wrote: Hi, Our potentiel next phone provider ask me a question i can't answer for sure, maybe someone here knows ? He says that is equipement only support VN4 protocol or more, or ETSI, however i can't find matching terms in the digium documentation or the chan_dahdi/dahdi/system.conf files... Those terms would be ISDN-related. VN4 is Version Number 4, and ETSI is the European standards-adopting organization for telecoms. So you might want to check for E1 support (ISDN in Europe, basically) if you want to connect a PRI-capable equipment - I assume that's what you are looking for since you mentioned the TE220. If you read French, you might want to look at this page also: http://blog.nicolargo.com/2008/01/installation-dune-carte-digium-avec-asterisk.html I found it very useful when I installed recently a TE220 card. Good luck, Laurent -- Laurent Steffan Consultant VOIP Web: http://VOIP.nc ___ -- 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] CRTC and FCC Feeds
I saw that already. It's not a listing of the FCCs headlines. It's just a very lame, unusable, unordered list of a few snippets from random FCC meetings. Check the data in my feed and compare it that and I think the answer to what about this becomes obvious. Cheers, Shidan On Fri, Jan 16, 2009 at 12:07 AM, Alex Balashov abalas...@evaristesys.com wrote: What about this? http://www.thefederalregister.com/rss/department/FEDERAL_COMMUNICATIONS_COMMISSION/ Shidan wrote: I don't understand why so many government sites fail to provide some sort of feed to their daily bulletins. What I am venting about in specific are the Canadian CRTC and FCC sites, every day I have to go to the website and when I reach the content, usually it isn't even HTML but a Word or PDF file. So finally today I decided to do something about it and wrote a little app that scrapes their daily releases and displays the information in blogger so I can just add the feeds to my reader. The CRTC site specially dissapoints me because someone who developed the site had the common sense of using Dublin Core in the meta tags or are using a CMS which obviously makes it easy to make available machine readable content. I wrote this system using Python, Beautiful Soup and Google's App. Engine, I will allow comments on both and if there is demand will switch to a PLIGG instance instead of blogger. As a legal note, I make absolutely no claims or warranties that this application actually works or the following blogs display accurate data from the CRTC or FCC's site. Here they are: for the CRTC: http://crtc.gulfpearl.com for the FCC: http://fcc.gulfpearl.com --- Shidan Gouran shidan.gulfpearl.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 -- 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 ___ -- 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] Digium TE220 supported protocol
Le 16.01.2009 04:11, Benoit a écrit : Hi, Our potentiel next phone provider ask me a question i can't answer for sure, maybe someone here knows ? He says that is equipement only support VN4 protocol or more, or ETSI, however i can't find matching terms in the digium documentation or the chan_dahdi/dahdi/system.conf files... Those terms would be ISDN-related. VN4 is Version Number 4, and ETSI is the European standards-adopting organization for telecoms. So you might want to check for E1 support (ISDN in Europe, basically) if you want to connect a PRI-capable equipment - I assume that's what you are looking for since you mentioned the TE220. If you read French, you might want to look at this page also: http://blog.nicolargo.com/2008/01/installation-dune-carte-digium-avec-asterisk.html I found it very useful when I installed recently a TE220 card. Good luck, Laurent -- Laurent Steffan Consultant VOIP Web: http://VOIP.nc ___ -- 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
