Re: [asterisk-users] no source on calllogs
just post your peer configs for one of your clients that don't show on the log. mostly it's IAX peers that don't show on the logs if not configured to. All my clients are sip peers actually. Here is the users.conf entry for one of the users that doesnt show on logs. [8006] username = 8006 transfer = yes mailbox = 8006 call-limit = 100 fullname = Test registersip = no host = dynamic callgroup = 1 call-limit = 100 context = DLPN_All cid_number = 8006 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = yes callwaiting = yes hasmanager = no hasagent = no hassip = yes hasiax = yes secret = nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 autoprov = no label = macaddress = linenumber = 1 LINEKEYS = 1 -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Tue, 28 Apr 2009 13:15:12 +0300 From: oguzh...@bilkent.edu.tr To: asterisk-users@lists.digium.com Subject: [asterisk-users] no source on calllogs Hello, As i check the call logs, some of my clients seem to make successful calls but, in logfiles, Source field seems empty..Still I can see who is the source from Channel tab as SIP/, and the called number and the time etc but.. nothing on Source and the Called ID tab. Just some clients has this problem. But as i check nothing special in their settings. What might cause this problem. Using Asterisk 1.6.0.9 (had the same problem with 1.6.0.6 too) Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live Hotmail®: more than just e-mail. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_more_042009___ -- Bandwidth and Colocation Provided 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] Cisco SPA525G
Anyone have used one of the new Cisco SPA525G with Asterisk ? Will be reading manual before starting to play with, but would really appreciate if you could share some tips with me. Thanks G. ___ -- Bandwidth and Colocation Provided 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] POS modems
Steve Underwood wrote: Hi, If anyone is interested in the low speed modems needed for POS applications (V.22, V.22bis, V.22bisFC and V.29FC) please contact me. I had some spare time while travelling, and finally got the V.22bis code I started a long time ago into a start where its basically functional. I'm now looking for input about exactly what application software expects from these modems, so I can plan the remainder of the code. Steve Does this mean that sipmodem is still on the back seat? ___ -- Bandwidth and Colocation Provided 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] finding the right amd.conf settings
Thanks very much. By the way, what exactly does silence threshold mean, how does it work, and what does the threshold value represent (bitrate? integer?)? The amd.conf and voip-info wiki doesn't describe it. On Tue, Apr 28, 2009 at 5:58 PM, Matt Riddell li...@venturevoip.com wrote: On 28/04/2009 7:27 p.m., Roi Stork wrote: My current amd.conf settings still allow a lot of answering machines to be undetected. This is the setup: [AnsweringMachineDetector] initial_silence = 2500 greeting = 1500 after_greeting_silence = 500 total_analysis_time = 3000 min_word_length = 120 between_words_silence = 50 maximum_number_of_words = 3 silence_threshold = 512 Asterisk version used is 1.2.24. I reduced the max number of words to 2, and I have to find out if it improves detection. Can anyone recommend a better setting? Your threshold is quite high - just make sure that words are being detected as words. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided 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] Replacement of Macro() with Gosub()
Hi,
Is there some more thorough documentation of this change that has
happened in 1.6? The upgrade.txt and changes.txt files mention it, but
I have already seen details of this change that do not appear to be
documented except in conversations on the mailing list...
1) It appears that it is no longer legal to have:
[macro-contaxtA]
...stuff...
[contextA]
...stuff...
Is this true? Or have I misunderstood the post that I read?
2) The single most useful feature of a macro was auto-return on
unmatched goto. This feature reduces the size of my dialplan to about
30% of its previous size! How can this be implemented with a Gosub?
eg.
[macro-specialcases]
exten = s,1,Goto(c-${ARG1})
exten = c-special,1,NoOp(I am special)
The above would execute the NoOp if ARG1 is set to special but just
return otherwise. Perhaps a Gosub can use the 'i' extension to do the
same?
[macro-specialcases]
exten = s,1,Goto(c-${ARG1})
exten = c-special,1,NoOp(I am special)
exten = c-special,2,Return
exten = i,2,Return
Perhaps?
Thanks for any pointers.
Regards,
Steve
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[asterisk-users] problem in upgrading to 1.6.1.0
Hello, I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in registering users. As i see from debug it successfully reads from users.conf but later,when a user tries to logon it say peer not found And there were an error msg about mysql about the username field..Smthing changed in mysql tables??? Now i downgraded to 1.6.0.9 again and everything is working.. ___ -- Bandwidth and Colocation Provided 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] Diference between volume of mp3 and wav files
Hi, I have some files in mp3 in my Asterisk but when I play it the volume is lo= wer than wav files. Both the files (wav and mp3) are encoded with the same = amplitude. In anothers players the audio volume of these files are equal. Can I fix this diference between volume of mp3 and wav file? Thanks Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.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] Diference between volume of mp3 and wav files
Check this link. http://www.voip-info.org/wiki/view/Asterisk+mpg123+faking+it Are you using mpg123? What version of Asterisk are you using? Why dont you just SOX the files into .gsm, .wav or some other Asterisk-happy format? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Enes Mateus Sent: Wednesday, April 29, 2009 8:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Diference between volume of mp3 and wav files Hi, I have some files in mp3 in my Asterisk but when I play it the volume is lo= wer than wav files. Both the files (wav and mp3) are encoded with the same = amplitude. In anothers players the audio volume of these files are equal. Can I fix this diference between volume of mp3 and wav file? Thanks _ Veja quais são os assuntos do momento no Yahoo! + Buscados: Top http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ 10 - Celebridades http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ celebridades/ - Música http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ m%C3%BAsica/ - Esportes http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ esportes/ ___ -- Bandwidth and Colocation Provided 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] AMD Not Working
Hi, Thanks for your reply. We donot kept any absolute time out's. And we have remove the AMD and kept only the play back, it works fine. Any help is highly appreciated. Thanks. On Wed, Apr 29, 2009 at 6:35 AM, Matt Riddell li...@venturevoip.com wrote: On 28/04/2009 4:56 p.m., Sam Hawkin wrote: Hi, Thanks for your reply. I have tried as you suggested. In h extension it is giving Status as AMD_HANGUP. That normally means that the remote end disconnected the call - if I were you I'd do a SIP debug to find out why the call is being disconnected. You don't have any absolute timeouts or anything? The other thing to test would be to skip AMD for the moment and just play some audio instead and see if it hangs up in that case. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided 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] Replacement of Macro() with Gosub()
On Wednesday 29 April 2009 07:27:44 Steve Davies wrote:
Hi,
Is there some more thorough documentation of this change that has
happened in 1.6? The upgrade.txt and changes.txt files mention it, but
I have already seen details of this change that do not appear to be
documented except in conversations on the mailing list...
1) It appears that it is no longer legal to have:
[macro-contaxtA]
...stuff...
[contextA]
...stuff...
Is this true? Or have I misunderstood the post that I read?
You've misunderstood. Gosub does not use the macro- prefix, and indeed,
there's no requirement that your gosub target even be a separate context. You
can have a gosub target be a high priority within your current extension (this
would probably be most appropriate when your extension is a wide-ranging
pattern match).
2) The single most useful feature of a macro was auto-return on
unmatched goto. This feature reduces the size of my dialplan to about
30% of its previous size! How can this be implemented with a Gosub?
eg.
[macro-specialcases]
exten = s,1,Goto(c-${ARG1})
exten = c-special,1,NoOp(I am special)
The above would execute the NoOp if ARG1 is set to special but just
return otherwise. Perhaps a Gosub can use the 'i' extension to do the
same?
It cannot. Your Goto must match something, and you will never return without
an explicit Return().
Let's also be clear about what Gosub is replacing. Gosub replaces Macro for
AEL2. The side effects of this are relatively unfelt, unless you're doing
something unusual like defining subroutines in AEL and calling them from
extensions.conf. The big gain in this is the ability to have infinite depths
of subroutines, as opposed to a maximum of about 7 in 1.4's AEL.
Macro is not going away. If you want to continue to use Macro, it's there for
you to use, warts and all. Macro continues to have the depth limit of 7
levels deep, and that won't change. It's a fundamental limit of the
application, which is one of the reasons why its functionality has been
replaced in AEL with Gosub. Also, I think Gosub is a bit easier to learn, and
its behavior is more straightforward. There aren't corner cases and special
behaviors that you have to learn about Gosub; it just works like a subroutine.
--
Tilghman
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Re: [asterisk-users] Diference between volume of mp3 and wav files
I'm using Asterisk 1.4.19 and asterisk-addons 1.4.7. So to play mp3 it uses format_mp3.so module. I use mp3 because I need play the same audio file in the web. --- Em qua, 29/4/09, Danny Nicholas da...@debsinc.com escreveu: De: Danny Nicholas da...@debsinc.com Assunto: Re: [asterisk-users] Diference between volume of mp3 and wav files Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Data: Quarta-feira, 29 de Abril de 2009, 10:53 Check this link. http://www.voip-info.org/wiki/view/Asterisk+mpg123+faking+it Are you using mpg123? What version of Asterisk are you using? Why don’t you just SOX the files into .gsm, .wav or some other “Asterisk-happy” format? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Enes Mateus Sent: Wednesday, April 29, 2009 8:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Diference between volume of mp3 and wav files Hi, I have some files in mp3 in my Asterisk but when I play it the volume is lo= wer than wav files. Both the files (wav and mp3) are encoded with the same = amplitude. In anothers players the audio volume of these files are equal. Can I fix this diference between volume of mp3 and wav file? Thanks Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10 - Celebridades - Música - Esportes -Anexo incorporado- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.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] I am looking for a good source of Monterrey DIDs
good point :) From: abalas...@evaristesys.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Wednesday, April 29, 2009 1:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] I am looking for a good source of Monterrey DIDs I cordially point you to asterisk-biz. -- Sent from mobile device On Apr 28, 2009, at 9:39 PM, Robert Augustyn robert.augus...@linqone.com wrote: Any pointers will be appreciated... Sincerely, Robert Augustyn ___ -- Bandwidth and Colocation Provided 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] Diference between volume of mp3 and wav files
Check this page out. http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf You might have to do some kind of custom-app tweak to play the mp3 at an increased volume. In my installation, when I take the gsm from voicemail, I have to SOX it to make it audible as an email attachment /usr/bin/sox msg1.WAV v 10 msg0001a.wav _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Enes Mateus Sent: Wednesday, April 29, 2009 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Diference between volume of mp3 and wav files I'm using Asterisk 1.4.19 and asterisk-addons 1.4.7. So to play mp3 it uses format_mp3.so module. I use mp3 because I need play the same audio file in the web. --- Em qua, 29/4/09, Danny Nicholas da...@debsinc.com escreveu: De: Danny Nicholas da...@debsinc.com Assunto: Re: [asterisk-users] Diference between volume of mp3 and wav files Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Data: Quarta-feira, 29 de Abril de 2009, 10:53 Check this link. http://www.voip-info.org/wiki/view/Asterisk+mpg123+faking+it http://www.voip-info.org/wiki/view/Asterisk+mpg123+faking+it Are you using mpg123? What version of Asterisk are you using? Why dont you just SOX the files into .gsm, .wav or some other Asterisk-happy format? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Enes Mateus Sent: Wednesday, April 29, 2009 8:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Diference between volume of mp3 and wav files Hi, I have some files in mp3 in my Asterisk but when I play it the volume is lo= wer than wav files. Both the files (wav and mp3) are encoded with the same = amplitude. In anothers players the audio volume of these files are equal. Can I fix this diference between volume of mp3 and wav file? Thanks _ Veja quais são os assuntos do momento no Yahoo! + Buscados: http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ Top 10 - http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ celebridades/ Celebridades - http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ m%C3%BAsica/ Música - http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ esportes/ Esportes -Anexo incorporado- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users _ Veja quais são os assuntos do momento no Yahoo! + Buscados: http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ Top 10 - http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ celebridades/ Celebridades - http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ m%C3%BAsica/ Música - http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ esportes/ Esportes ___ -- Bandwidth and Colocation Provided 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] Replacement of Macro() with Gosub()
2009/4/29 Tilghman Lesher tilgh...@mail.jeffandtilghman.com: Let's also be clear about what Gosub is replacing. Gosub replaces Macro for AEL2. The side effects of this are relatively unfelt, unless you're doing something unusual like defining subroutines in AEL and calling them from extensions.conf. The big gain in this is the ability to have infinite depths of subroutines, as opposed to a maximum of about 7 in 1.4's AEL. Macro is not going away. If you want to continue to use Macro, it's there for you to use, warts and all. Macro continues to have the depth limit of 7 levels deep, and that won't change. It's a fundamental limit of the application, which is one of the reasons why its functionality has been replaced in AEL with Gosub. Also, I think Gosub is a bit easier to learn, and its behavior is more straightforward. There aren't corner cases and special behaviors that you have to learn about Gosub; it just works like a subroutine. Many thanks for the clarification - I do use both Macro() and Gosub() at present, and understand the difference ( I try to tell myself I do ;-) ). I do not use AEL dialplans. I had misunderstood the description on the changelog, and I had not noticed that it was an AEL change, rather than a generic dialplan change. There is clearly more of a difference between extensions.ael usage and extensions.conf usage than I realised. I can see now that AEL2 parses Macro () as a builtin, whereas extensions.conf will treat it as an application, and they have little in common. Cheers, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Something wrong with DAHDI signalling according to the CLI
I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO modules. When I plug one PSTN-line into a FXO-port I am able to receive calls on this line and I can also make calls from an internal SIP-phone to the external PSTN-network. Still I am bothered about something that appears on the CLI when I do a reload chan_dahdi.so : asterisk*CLI reload chan_dahdi.so -- Reloading module 'chan_dahdi' (DAHDI Telephony w/PRI) == Parsing '/etc/asterisk/chan_dahdi.conf': Found [Apr 29 16:26:53] WARNING[8927]: chan_dahdi.c:11951 process_dahdi: Ignoring any changes to 'signalling' (on reload) [Apr 29 16:26:53] WARNING[8927]: chan_dahdi.c:4090 handle_alarms: Detected alarm on channel 1: Red Alarm -- Reconfigured channel 1, FXS Kewlstart signalling [Apr 29 16:26:53] WARNING[8927]: chan_dahdi.c:11951 process_dahdi: Ignoring any changes to 'signalling' (on reload) [Apr 29 16:26:53] WARNING[8927]: chan_dahdi.c:4090 handle_alarms: Detected alarm on channel 2: Red Alarm -- Reconfigured channel 2, FXS Kewlstart signalling [Apr 29 16:26:53] WARNING[8927]: chan_dahdi.c:11951 process_dahdi: Ignoring any changes to 'signalling' (on reload) -- Reconfigured channel 3, FXS Kewlstart signalling == Parsing '/etc/asterisk/users.conf': Found What about this ignoring any changes to 'signalling' ??? It's not a notice, its really a warning... I would gladly see no warning at all when reloading DAHDI. This is my chan_dahdi.conf (created manually) : [trunkgroups] [channels] language=be busydetect=yes busycount=6 callerid=asreceived ;threewaycaling=yes transfer=yes ; Note that if any of your DAHDI cards have hardware echo cancellers, ; then this setting only turns them on and off. There are no special ; settings required for hardware echo cancellers; when present and ; enabled in their kernel modules, they take precedence over the ; software echo canceller compiled into DAHDI automatically. echocancel=yes ;internationalprefix: 00 ;nationalprefix:0 context=tel2268191 ; E.U. signalling=fxs_ks group=1 channel = 1 context=tel2362334 ; KMO signalling=fxs_ks group=1 channel = 2 context=telunknown signalling=fxs_ks group=1 channel = 3 This is what DAHDI config tells me : [r...@asterisk asterisk]# /usr/sbin/dahdi_cfg -vvv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) 3 channels to configure. It keeps telling me that the channels need to be configured, in stead of saying that my channels ARE configured. Thanks for the feedback ! Jonas. ___ -- Bandwidth and Colocation Provided 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 sudden crash
Hi I am using asterisk-1.6.0.6 and I have noticed strange behaviour lately. When a user ends his call asterisk executes twice the h extensions (in my case this is the AGI script) and writes this to the logs: cdr.c: CDR on channel 'SIP/xx-b6623038' already posted. and after that it crashes immediately. This had happened twice so far. Does anyone know what is causing this.? Cheers Andrew ___ -- Bandwidth and Colocation Provided 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] Replacement of Macro() with Gosub()
On Wednesday 29 April 2009 11:45:13 Steve Davies wrote: 2009/4/29 Tilghman Lesher tilgh...@mail.jeffandtilghman.com: Let's also be clear about what Gosub is replacing. Gosub replaces Macro for AEL2. The side effects of this are relatively unfelt, unless you're doing something unusual like defining subroutines in AEL and calling them from extensions.conf. The big gain in this is the ability to have infinite depths of subroutines, as opposed to a maximum of about 7 in 1.4's AEL. Macro is not going away. If you want to continue to use Macro, it's there for you to use, warts and all. Macro continues to have the depth limit of 7 levels deep, and that won't change. It's a fundamental limit of the application, which is one of the reasons why its functionality has been replaced in AEL with Gosub. Also, I think Gosub is a bit easier to learn, and its behavior is more straightforward. There aren't corner cases and special behaviors that you have to learn about Gosub; it just works like a subroutine. Many thanks for the clarification - I do use both Macro() and Gosub() at present, and understand the difference ( I try to tell myself I do ;-) ). I do not use AEL dialplans. I had misunderstood the description on the changelog, and I had not noticed that it was an AEL change, rather than a generic dialplan change. There is clearly more of a difference between extensions.ael usage and extensions.conf usage than I realised. I can see now that AEL2 parses Macro () as a builtin, whereas extensions.conf will treat it as an application, and they have little in common. Well, Macro is officially deprecated, but not in the it's going away sense of deprecated, merely in the way that we encourage new users to use Gosub, when they implement dialplan subroutines. -- 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] Replacement of Macro() with Gosub()
A couple of days ago we decide to test 1.6 branch, we are using 1.4 in our production environments, and to my surprise, when ported parts of our dialplan, I found that any macro declaration in the AEL dialplan, we use them a lot for subroutines, was ported as gosub when compiled. We are now checking the rest of the changes made in 1.6, but this one it quite useful if any of you do AEL. LTenorio Tilghman Lesher wrote: On Wednesday 29 April 2009 11:45:13 Steve Davies wrote: 2009/4/29 Tilghman Lesher tilgh...@mail.jeffandtilghman.com: Let's also be clear about what Gosub is replacing. Gosub replaces Macro for AEL2. The side effects of this are relatively unfelt, unless you're doing something unusual like defining subroutines in AEL and calling them from extensions.conf. The big gain in this is the ability to have infinite depths of subroutines, as opposed to a maximum of about 7 in 1.4's AEL. Macro is not going away. If you want to continue to use Macro, it's there for you to use, warts and all. Macro continues to have the depth limit of 7 levels deep, and that won't change. It's a fundamental limit of the application, which is one of the reasons why its functionality has been replaced in AEL with Gosub. Also, I think Gosub is a bit easier to learn, and its behavior is more straightforward. There aren't corner cases and special behaviors that you have to learn about Gosub; it just works like a subroutine. Many thanks for the clarification - I do use both Macro() and Gosub() at present, and understand the difference ( I try to tell myself I do ;-) ). I do not use AEL dialplans. I had misunderstood the description on the changelog, and I had not noticed that it was an AEL change, rather than a generic dialplan change. There is clearly more of a difference between extensions.ael usage and extensions.conf usage than I realised. I can see now that AEL2 parses Macro () as a builtin, whereas extensions.conf will treat it as an application, and they have little in common. Well, Macro is officially deprecated, but not in the it's going away sense of deprecated, merely in the way that we encourage new users to use Gosub, when they implement dialplan subroutines. ___ -- Bandwidth and Colocation Provided 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] Verifone-Asterisk-AGI
Wrong list. asterisk-dev is for changing the C source code of Asterisk. That's part of why you didn't get a response yesterday. On Wed, 29 Apr 2009, Juan Miguel Quiros Arrieta wrote: I have to develop an application using the VeriFone vx510 device and I read this device needed or could use a PPPoE connection in order to validate and send all information collected from the end user. My question is if I can use the asterisk and an IVR development using AGI to interact with the VeriFone. I mean, VeriFone-E1 or FXs-o-Asterisk-MyAgiIVR. And if its possible to send information in real time to the VeriFone. Has someone done this. A little googling yields some information that may help. The VeriFone vx510 is a credit card terminal. It has Ethernet and 12mb of memory. There are different models with different capabilities based on country of origin so you should state where you plan to deploy the device. I didn't see any programming manuals VerifFone's web site, but I didn't look for but a few seconds. I'm guessing the connection between this device and whatever will be encrypted and obscured by some proprietary protocol that will be economically impossible without full documentation. If you are asking if someone can code an AGI to take information possibly passed in channel variables, pass this information (an authorization or sale request) to the vx510 via a TCP/UDP connection, receive a response and return the response to your dialplan via channel variables, the answer is yes -- assuming VeriFone's documentation is helpful. 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
Re: [asterisk-users] Bounty for parking on slot@context
Wrong list. asterisk-dev is for changing the C source code of Asterisk. I don't think AGI's count or are considered for inclusion into the subversion repository as stated by one of your conditions for payment. On Wed, 29 Apr 2009, Alistair Cunningham wrote: I'd like to offer a bounty for a feature for Asterisk where an AGI program can park and retrieve calls on parking slots that are of the form slot@context, where slot is a number and context is an arbitrary string. Asterisk will not choose a slot; the AGI program will always specify which exact slot and context to use. The AGI will keep track of which to use in its own database. It's acceptable for Asterisk to return an error if the AGI tries to park on a slot that's already in use. If you want to implement automatic choosing of slots for the benefit of other Asterisk users that's fine, but our code will always specify which slot to use and your code must at least have an option to allow this. It must not be necessary to pre-define either slots or contexts in Asterisk configuration files. It must be completely dynamic for the AGI to choose at run-time. The call flow should work as follows: 1. A call comes in from an external number. 2. Asterisk passes control to the AGI. 3. The AGI does a Dial() with the t option to a SIP phone on a remote SIP registrar. 4. The destination transfers the call using # or its own transfer button. The # for the transfer should be the only thing in this call flow that's set in the Asterisk configuration files. 5. The transfer call comes back to the AGI, which decides that the transferrer wants to park the call. Each transferring phone may have a different code to park calls, so this is configured in the AGI's database rather than Asterisk's features.conf. 6. The AGI instructs Asterisk to park the original inbound call on slot@context. 7. Asterisk does this, and reads slot to the transferring phone. 8. The transferring phone hangs up. 9. Another telephone calls a number to pick up the parked call that goes to the AGI. This number may be slot or it may not. For example, a user may have *99slot configured in the AGI's database to pick up parked calls. Only the AGI should care about this. 10. The AGI instructs Asterisk to connect this call with the call parked on slot@context. Steps 1, 2, 3, 4, 5, 8, and 9 already exist. Steps 6, 7, and 10 need written. I'm also open to other ideas for call flows; please discuss with me before starting work however. If anyone would like to write this, and it gets accepted into the Asterisk subversion repository for a future Asterisk version, Integrics is willing to pay a bounty of USD 500, payable by PayPal. I think you need another 0 in the bounty. 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
Re: [asterisk-users] dahdi_dummy: Unable to register DAHDI rtc driver
On Wed, Apr 8, 2009 at 10:23 AM, Shaun Ruffell sruff...@digium.com wrote: David Backeberg wrote: Hello there: I think I have a silly kernel configuration problem. I'm running: * vanilla 2.6.27.10 kernel built from source * dahdi-2.1.0.4 built from source So far so good, dahdi module loads just fine: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.4 when I try to: hal04 dahdi # modprobe dahdi_dummy FATAL: Error inserting dahdi_dummy (/lib/modules/2.6.27.10/dahdi/dahdi_dummy.ko): Input/output error kernel messages gives me: dahdi_dummy: Unable to register DAHDI rtc driver There is a kernel config parameter called CONFIG_HPET_EMULATE_RTC, that if defined in the kernel config, will cause the behavior that you're seeing. I have not looked to see at which version that option came in. However, I think support for RTC in dahdi_dummy is best dropped. There is a patch on mantis (http://bugs.digium.com/view.php?id=13930) against dahdi_dummy that allows it to provide accurate timing with standard kernel timers that you may want to try. I applied the patch on Bug 13930, tested it out and it has been a great success. I now see that the Bug has been closed and the patch has been added to DAHDI trunk. Hooray! As a piece of documentation, I now get a different report from dahdi_tool when using dahdi_dummy. dahdi_tool used to report that the source was RTC. Now the source is Linux26. The dahdi_test -c 20 results were much more consistent with Bug 13930's patch than when using RTC as the timing source using my 2.6.25.9 kernel. ___ -- Bandwidth and Colocation Provided 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 sudden crash
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andrew Nowrot wrote: This had happened twice so far. Does anyone know what is causing this.? Start by upgrading to 1.6.0.9, then if it continues you can start tracking it down. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ+KjICFu3bIiwtTARAgiQAJ0WEGJNroqfnfpQWnUABt4Uh4KguQCgrE23 sGSrNv9+nKyaT08PC0Izc0g= =e5bf -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided 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] What do I need to connect landline calls without telephony hardware?
For some reason, I have been unable to find the answer to this online or in books... I want to have a click-to-connect feature on my website where the user enters their phone number and then my asterisk server calls their phone and my phone and connects the two calls to each other. All I have are: 1. A Server 2. A DSL connection 3. A Router and DSL Modem 4. A static IP What do i need to be able to do this? Can I just install asterisk and it will work, or do I have to pay some company for a gateway or something? What do i require beyond the asterisk server? Thank you for any help you guys can give! Don ___ -- Bandwidth and Colocation Provided 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] no source on calllogs
try adding callerid=CIDNAME CIDNUM this will force your callerID in your DIalplan -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Wed, 29 Apr 2009 09:38:58 +0300 From: oguzh...@bilkent.edu.tr To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] no source on calllogs just post your peer configs for one of your clients that don't show on the log. mostly it's IAX peers that don't show on the logs if not configured to. All my clients are sip peers actually. Here is the users.conf entry for one of the users that doesnt show on logs. [8006] username = 8006 transfer = yes mailbox = 8006 call-limit = 100 fullname = Test registersip = no host = dynamic callgroup = 1 call-limit = 100 context = DLPN_All cid_number = 8006 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = yes callwaiting = yes hasmanager = no hasagent = no hassip = yes hasiax = yes secret = nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 autoprov = no label = macaddress = linenumber = 1 LINEKEYS = 1 -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Tue, 28 Apr 2009 13:15:12 +0300 From: oguzh...@bilkent.edu.tr To: asterisk-users@lists.digium.com Subject: [asterisk-users] no source on calllogs Hello, As i check the call logs, some of my clients seem to make successful calls but, in logfiles, Source field seems empty..Still I can see who is the source from Channel tab as SIP/, and the called number and the time etc but.. nothing on Source and the Called ID tab. Just some clients has this problem. But as i check nothing special in their settings. What might cause this problem. Using Asterisk 1.6.0.9 (had the same problem with 1.6.0.6 too) Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live™ Hotmail®:…more than just e-mail. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_more_042009___ -- Bandwidth and Colocation Provided 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 _ Windows Live™ Hotmail®:…more than just e-mail. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_more_042009___ -- Bandwidth and Colocation Provided 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 do I need to connect landline calls without telephony hardware?
On Wednesday 29 April 2009 14:14:58 don rhummy wrote: For some reason, I have been unable to find the answer to this online or in books... I want to have a click-to-connect feature on my website where the user enters their phone number and then my asterisk server calls their phone and my phone and connects the two calls to each other. All I have are: 1. A Server 2. A DSL connection 3. A Router and DSL Modem 4. A static IP What do i need to be able to do this? Can I just install asterisk and it will work, or do I have to pay some company for a gateway or something? What do i require beyond the asterisk server? You need a provider who will allow at least 2 concurrent calls through her gateway. One call is to the user; the other call is to you. The mechanism for doing this is the Originate function. You can originate through the CLI, through a manager connection, or by placing a callfile in the call spool directory. -- 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] Something wrong with DAHDI signalling according to the CLI
On Wed, Apr 29, 2009 at 07:41:14PM +0200, jonas kellens wrote: I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO modules. When I plug one PSTN-line into a FXO-port I am able to receive calls on this line and I can also make calls from an internal SIP-phone to the external PSTN-network. Still I am bothered about something that appears on the CLI when I do a reload chan_dahdi.so : asterisk*CLI reload chan_dahdi.so -- Reloading module 'chan_dahdi' (DAHDI Telephony w/PRI) == Parsing '/etc/asterisk/chan_dahdi.conf': Found [Apr 29 16:26:53] WARNING[8927]: chan_dahdi.c:11951 process_dahdi: Ignoring any changes to 'signalling' (on reload) chan_dahdi does not add channels, remove channels or changes the signalling of channels (as well as most configurations of PRI spans) on reload. Either restart Asterisk or use 'dahdi restart'. Both methods will disconnect all existing DAHDI calls. -- 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] Asterisk 1.6 and CDR/MySQL
Thank you .. appreciated. Best Regards, -- SplatNIX IT Services :: Innovation through collaboration - Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Tuesday 28 April 2009 15:35:14 --[ UxBoD ]-- wrote: HI, I am trying to setup CDR with ODBC and MySQL but get the following error :- [Apr 28 21:30:01] ERROR[14567]: cdr_odbc.c:133 odbc_log: Unable to retrieve database handle. CDR failed. I can successfully connect with iSQL so ODBCINST and ODBC ini files must be okay. I have modified /etc/asterisk/cdr_odbc.conf to include :- [global] dsn=asterisk username=asterisk password=*** ;loguniqueid=yes ;dispositionstring=yes table=cdr ;cdr is default table name usegmtime=yes ; set to yes to log in GMT What am I doing wrong please ? If you read UPGRADE.txt, you'd know that cdr_odbc.conf now refers to pooled connections in res_odbc.conf. It's an extra layer of indirection, but it ensures that all ODBC connections in Asterisk are using the same set of pooled connections with error correction and whatnot. ___ -- Bandwidth and Colocation Provided 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] Cisco SPA525G
On Apr 28, 2009, at 11:36 PM, Gondar Monn wrote: Anyone have used one of the new Cisco SPA525G with Asterisk ? Will be reading manual before starting to play with, but would really appreciate if you could share some tips with me. Thanks We tested one a few months ago. They work like the other SPA series phones. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk sudden crash
OK I will do that. i let you know about the results. Cheers On Wed, Apr 29, 2009 at 9:21 PM, Barry L. Kline blkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andrew Nowrot wrote: This had happened twice so far. Does anyone know what is causing this.? Start by upgrading to 1.6.0.9, then if it continues you can start tracking it down. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ+KjICFu3bIiwtTARAgiQAJ0WEGJNroqfnfpQWnUABt4Uh4KguQCgrE23 sGSrNv9+nKyaT08PC0Izc0g= =e5bf -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided 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] US Caller ID
Okay, I can't find what might be causing this. Here is what I got: Asterisk server hooked up to a digital T1 line (full 24-channel) via a Digium card. Verizon has turned on caller ID on the first line (I can guarantee it is on as I can hear the FSK tones on this line but not the others). Using zttool an ZapScan() I have determined the following: 1) The RxB/RxD bits toggle from 1 to 0 signaling a ring. 2) A short time later, via ZapScan() I can hear the FSK tone. 3) About the same time I hear the FSK tone I see the Starting simple switch line in the Asterisk console. 4) Next I see the second ring trigger in zttool and then Asterisk say ss_thread: Got event 18 (Ring Begin). Caller ID never shows up. I have tried cranking the rxgain up thinking maybe it was too quiet for Asterisk to detect but that did not help. My caller id settings in zapata.conf are: usecallerid=yes callerid=asreceived cidsignalling=bell cidstart=ring signalling=fxs_ks Is there any existing debug options I can turn on, or do I need to add some to try and figure out what is going on; or does somebody have an instant answer for me? Thanks, Daniel ___ -- Bandwidth and Colocation Provided 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] AMD Not Working
On 30/04/2009 2:25 a.m., Sam Hawkin wrote: Hi, Thanks for your reply. We donot kept any absolute time out's. And we have remove the AMD and kept only the play back, it works fine. Any help is highly appreciated. Ok, so when you remove AMD and keep playback, how long is the message. Secondly, how long does it take before you are disconnected with AMD. Oh, and which version of Asterisk are you running? -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided 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] Something wrong with DAHDI signalling according to the CLI
On 30/04/2009 5:41 a.m., jonas kellens wrote: I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO modules. [*Apr 29 16:26:53] WARNING[8927]: chan_dahdi.c:11951 process_dahdi: Ignoring any changes to 'signalling' (on reload) * Changes to signalling are not used when you reload - the warning tells you about this. I.E. if you changed the signalling to PRI_NET or whatever, you would need to restart rather than reload - maybe it should be a notice rather than a warning. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided 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] finding the right amd.conf settings
On 29/04/2009 9:06 p.m., Roi Stork wrote: Thanks very much. By the way, what exactly does silence threshold mean, how does it work, and what does the threshold value represent (bitrate? integer?)? The amd.conf and voip-info wiki doesn't describe it. Basically when you speak, silence threshold is how loud silence is. On a mobile call there might be quite a bit of background noise, so the silence threshold needs to be a bit higher. If you have noise that is louder than the threshold then it will be considered as speaking. Conversely, if you have words that are quieter than the threshold then they will be considered as silence. I'm not sure what the units are, doesn't really make much difference. Basically you want it to be as low as possible, but not so low that it recognizes background noise as words. Just have a play round with it while watching the console - call cellphones, answer machines, IVRs etc and keep tuning it till you get the results you want. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided 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 do I need to connect landline calls without telephony hardware?
On 30/04/2009 7:14 a.m., don rhummy wrote: For some reason, I have been unable to find the answer to this online or in books... I want to have a click-to-connect feature on my website where the user enters their phone number and then my asterisk server calls their phone and my phone and connects the two calls to each other. All I have are: 1. A Server 2. A DSL connection 3. A Router and DSL Modem 4. A static IP What do i need to be able to do this? Can I just install asterisk and it will work, or do I have to pay some company for a gateway or something? What do i require beyond the asterisk server? http://lmgtfy.com/?q=asterisk+click+to+call :) -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided 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] 2nd Parking Lot
Does anybody know of a way to make another parking lot for version 1.2? We have a multi-tenant setup and it is set for x700 for parking. Well we added some new users and not thinking, we assigned them x700. I can't change the parking number as it will mess up the other users and the new user with x700 doesn't want to change. I was hoping there was some trickery that I can do to create a new (or another) parking lot, but I can't figure it out. ___ -- Bandwidth and Colocation Provided 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 do I need to connect landline calls without telephony hardware?
Um, no... I want to implement it myself. My question is with regard to asterisk, and getting it top actually make calls, etc, what do i need to make those outgoing and connecting calls with asterisk? --- On Wed, 4/29/09, Matt Riddell li...@venturevoip.com wrote: From: Matt Riddell li...@venturevoip.com Subject: Re: [asterisk-users] What do I need to connect landline calls without telephony hardware? To: donrhu...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, April 29, 2009, 5:40 PM On 30/04/2009 7:14 a.m., don rhummy wrote: For some reason, I have been unable to find the answer to this online or in books... I want to have a click-to-connect feature on my website where the user enters their phone number and then my asterisk server calls their phone and my phone and connects the two calls to each other. All I have are: 1. A Server 2. A DSL connection 3. A Router and DSL Modem 4. A static IP What do i need to be able to do this? Can I just install asterisk and it will work, or do I have to pay some company for a gateway or something? What do i require beyond the asterisk server? http://lmgtfy.com/?q=asterisk+click+to+call :) -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided 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 do I need to connect landline calls without telephony hardware?
On 30/04/2009 11:00 a.m., don rhummy wrote: Um, no... I want to implement it myself. My question is with regard to asterisk, and getting it top actually make calls, etc, what do i need to make those outgoing and connecting calls with asterisk? Did you go to the link I sent you? It was to a page of ways to do it. One of the links is: http://www.voipuser.org/forum_topic_9971.html Basically you'll need to make a php page that either connects to the Asterisk Manager, or creates a .call file and moves it to /var/spool/asterisk/outgoing -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided 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] Bounty for parking on slot@context
You're saying this is worth $5k ? This can be done in 2-3 hrs so are you really charging $1666-2500 an hour ? Martin On Wed, Apr 29, 2009 at 1:32 PM, Steve Edwards asterisk@sedwards.com wrote: If anyone would like to write this, and it gets accepted into the Asterisk subversion repository for a future Asterisk version, Integrics is willing to pay a bounty of USD 500, payable by PayPal. I think you need another 0 in the bounty. 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
Re: [asterisk-users] Bounty for parking on slot@context
Un-top-posting... On Wed, Apr 29, 2009 at 1:32 PM, Steve Edwards asterisk@sedwards.com wrote: On Wed, 29 Apr 2009, Alistair Cunningham wrote: If anyone would like to write this, and it gets accepted into the Asterisk subversion repository for a future Asterisk version, Integrics is willing to pay a bounty of USD 500, payable by PayPal. I think you need another 0 in the bounty. On Wed, 29 Apr 2009, Martin wrote: You're saying this is worth $5k ? This can be done in 2-3 hrs so are you really charging $1666-2500 an hour ? Why, yes. Aren't you? Seriously though, I'd probably piss away a couple of hours emailing back and forth nailing down the exact functional requirements, criteria for acceptance, exploring capacity needs, suggesting solutions, agreeing on documentation requirements, negotiating support expectations, developing functional testing, load testing, customer review, reworking, debugging, and accommodating the inevitable misunderstandings especially since we speak such different languages (English versus American). That sort of stuff (and a bunch more that didn't spontaneously spill out of my it's after 6pm alcohol induced stupor). Oh. And then I'd have to add in a couple of hours to actually code it. 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
[asterisk-users] ExtenSpy d option 1.6
I just installed asterisk 1.6.0.9 in the hope of using the d option in ExtenSpy(). I found this on the internet: - w - Enable 'whisper' mode, so the spying channel can talk to - the spied-on channel. - W - Enable 'private whisper' mode, so the spying channel can - talk to the spied-on channel but cannot listen to that - channel. + b - Only spy on channels involved in a bridged call. + B - Instead of whispering on a single channel barge in on both + channels involved in the call. + d - Override the typical numeric DTMF functionality and instead + use DTMF to switch between spy modes. + 4 = spy mode + 5 = whisper mode + 6 = barge mode + g(grp) - Only spy on channels in which one or more of the groups + listed in 'grp' matches one or more groups from the + SPYGROUP variable set on the channel to be spied upon. + Note that both 'grp' and SPYGROUP can contain either a + single group or a colon-delimited list of groups, such + as 'sales:support:accounting'. http://www.das-asterisk-buch.de/2.1/applications-extenspy.html When I do a core show application extenspy the d option does not show up. Anyone know about this d option how to use it ? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automon *1 not working; asterisk-1.4.22.1
automon is not working for me with asterisk 1.4.22.1 in extension.conf [globals] DYNAMIC_FEATURES=automon dial is with w feature.conf automon = *1 -- Executing [...@internal:1] Playback(SIP/218-007556b0, transfer) in new stack -- SIP/218-007556b0 Playing 'transfer' (language 'en') -- Executing [...@internal:2] Dial(SIP/218-007556b0, SIP/11IAX2/iaxy-322|30|rw) in new stack -- Called 11 -- Called iaxy-322 -- Call accepted by 10.0.0.108 (format ulaw) -- Format for call is ulaw -- SIP/11-007ccf50 is ringing -- IAX2/iaxy-322-15131 is ringing -- SIP/11-007ccf50 answered SIP/218-007556b0 -- Hungup 'IAX2/iaxy-322-15131 pressing *1 doesn't do anything. -- Joseph ___ -- Bandwidth and Colocation Provided 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] Bounty for parking on slot@context
No more questions. This all can be done in 2-3 hrs [PERIOD]. Martin On Wed, Apr 29, 2009 at 8:02 PM, Steve Edwards asterisk@sedwards.com wrote: Un-top-posting... On Wed, Apr 29, 2009 at 1:32 PM, Steve Edwards asterisk@sedwards.com wrote: On Wed, 29 Apr 2009, Alistair Cunningham wrote: If anyone would like to write this, and it gets accepted into the Asterisk subversion repository for a future Asterisk version, Integrics is willing to pay a bounty of USD 500, payable by PayPal. I think you need another 0 in the bounty. On Wed, 29 Apr 2009, Martin wrote: You're saying this is worth $5k ? This can be done in 2-3 hrs so are you really charging $1666-2500 an hour ? Why, yes. Aren't you? Seriously though, I'd probably piss away a couple of hours emailing back and forth nailing down the exact functional requirements, criteria for acceptance, exploring capacity needs, suggesting solutions, agreeing on documentation requirements, negotiating support expectations, developing functional testing, load testing, customer review, reworking, debugging, and accommodating the inevitable misunderstandings especially since we speak such different languages (English versus American). That sort of stuff (and a bunch more that didn't spontaneously spill out of my it's after 6pm alcohol induced stupor). Oh. And then I'd have to add in a couple of hours to actually code it. 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 ___ -- Bandwidth and Colocation Provided 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] valetparking.c
I found a version of valetparking.c that looks like what I need for adding more parking. However when I try and compile it in 1.2, it is looking for a file parking.h, which isn't in my install. I even looked through CVS and 1.0 and 1.2 to make sure it isn't an old file and I can't find it anywhere. Does anybody have an idea where I might get parking.h? Or what should be in it? Or is there a newer better version of app_valetparking.c? Thanks. Peder ___ -- Bandwidth and Colocation Provided 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] AMD Not Working
Hi, Thanks for your reply. We I remove the AMD it plays the message in the 12 seconds. It takes 16 seconds before AMD disconnects. We are using Asterisk 1.2.4 Any help is highly appreciated. Thanks. On Thu, Apr 30, 2009 at 3:00 AM, Matt Riddell li...@venturevoip.com wrote: On 30/04/2009 2:25 a.m., Sam Hawkin wrote: Hi, Thanks for your reply. We donot kept any absolute time out's. And we have remove the AMD and kept only the play back, it works fine. Any help is highly appreciated. Ok, so when you remove AMD and keep playback, how long is the message. Secondly, how long does it take before you are disconnected with AMD. Oh, and which version of Asterisk are you running? -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided 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] AMD Not Working
On 30/04/2009 4:26 p.m., Sam Hawkin wrote: Hi, Thanks for your reply. We I remove the AMD it plays the message in the 12 seconds. It takes 16 seconds before AMD disconnects. We are using Asterisk 1.2.4 Any help is highly appreciated. Few things: 1. Play the message twice without AMD (you might be being disconnected after 15 seconds) 2. I thought AMD wasn't present in 1.2. Is it a backport? 3. 1.2.4 is quite an old version, any chance you could upgrade it to a more recent version? There have been many bugs fixed since 1.2.4 was released. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
