Re: [asterisk-users] music on hold file formats
On Tue, 23 Jun 2009, Ron wrote: Hi, what software do i need to convert an mp3 to a g729 format? I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is being used by the user. Have you tried the convert command inside Asterisk? (1.4+) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] internal_timing not working (re: SIP silence suppression)
David Backeberg dbackeb...@gmail.com writes: As I understand it, you have enabled silence suppression, the silence is getting suppressed, and that is worse than when you were not using silence suppression. So how about not using silence suppression? Silence suppression isn't just a break your telephony option, it actually has a purpose. It can often save half your RTP bandwidth at close to zero cost in quality. It would be nice if it worked well in Asterisk. (Not that I can say personally whether it works; it's been 3 years since I tried it). /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
[asterisk-users] problem with sangoma card a108d
Hello, I need help to use my sangoma card a108d. I need that another server give me an E1 with a clock. The server with the sangoma reseive the E1 clock on port1 and is MASTER E1 on port2. But, I cant receive the clock (I am connected). Anyone can help me? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided 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 + Jabber
I want to use JabberSend in my dialplan, but I saw that my Asterisk does not support Jabber. Also I have nowhere a module res_jabber.so... So I thought I'd rebuild my Asterisk. In menuselect I saw that res_jabber was dependent of 'iksemel' and 'gnutls'. In my yum repositories I can find a gnutls.i386, but what is this iksemel-beast ??? There is info to find via google on the configuration of jabber.conf and the integration of jabber asterisk. But I can not find info on how to build jabber-support with Asterisk. I assume you do this via menuselect. So, what about this 'iksemel' ?? 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
Re: [asterisk-users] asterisk and google talk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Michael Maxwell wrote: I had the same issue, only from gtalk to asterisk with some connections and not others.. Asterisk to gtalk works fine here. Have you allowed the RTP ports past your firewall? small update ... with * 1.6.1.1 : - google talk (official client) to asterisk : no sound - xmpp jingle (on nokia n810) to asterisk : sound ok - xmpp jingle (on nokia n810) to google talk (official client) : sound ok this problem is only present with the google galk official client and asterisk ... firewall is a linux-box with no filtering ... asterisk, nokia n810 and google talk are behind the same router. - -- antoine patte -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkpB+mkACgkQBnIOcv+j7+zjtQCg7woQ/8teuCMAptJVBKLwEsi8 ItAAn1mAgTfeO0spD9GC0ueRqQFP8rEn =cxSh -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] Message Waiting Indication Astersk and kamailio
hi all, I have Asterisk 1.6.0.5 Installed and kamailio 1.0.5 version installed when i leave voicemail On Asterisk i need MWI Indication on kamailio extension there are some methods i tried but still cant get success All other feature are working fine also try voip-info.org methods can anybody suggest me for different method and have some different setting on SIP . any help appreciated regards Dhaval ___ -- Bandwidth and Colocation Provided 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] Outgoing CallerID for KPN in Belgium
Hi,
I'm using a ISDN-30 E1 line from KPN Belgium.
The challenge is to get a correct CallerID on outgoing lines.
When I put this in my dialplan:
exten = _0.,1,Set(TEMPVAR=${CALLERID(num):1})
exten = _0.,2,Set(CALLERID(num)=144622${TEMPVAR})
exten = _0.,3,NoOp(${CALLERID(num)})
exten = _0.,4,Dial(Zap/g1/${EXTEN:1},,)
The resulting CallerID is accepted by the telco, but on phones it shows for
instance as:
+14462241, whereas it should be +3214462241. So it seems the telco adds a +.
I've tried to then use:
exten = _0.,2,Set(CALLERID(num)=32144622${TEMPVAR})
but the telco seems not to accept this since it sends the general CallerID out.
Any clues on what I need to change to get this working? Is it something in
zapata.conf? Is it related to nationalprefix and internationalprefix?
Thank you!
B.
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Re: [asterisk-users] Outgoing CallerID for KPN in Belgium
Bart Coninckx wrote:
Hi,
I'm using a ISDN-30 E1 line from KPN Belgium.
The challenge is to get a correct CallerID on outgoing lines.
When I put this in my dialplan:
exten = _0.,1,Set(TEMPVAR=${CALLERID(num):1})
exten = _0.,2,Set(CALLERID(num)=144622${TEMPVAR})
exten = _0.,3,NoOp(${CALLERID(num)})
exten = _0.,4,Dial(Zap/g1/${EXTEN:1},,)
The resulting CallerID is accepted by the telco, but on phones it shows for
instance as:
+14462241, whereas it should be +3214462241. So it seems the telco adds a
+. I've tried to then use:
exten = _0.,2,Set(CALLERID(num)=32144622${TEMPVAR})
but the telco seems not to accept this since it sends the general CallerID
out.
Any clues on what I need to change to get this working? Is it something in
zapata.conf? Is it related to nationalprefix and internationalprefix?
Your best bet is to ask your telco what caller ID format you should be
presenting. Asterisk is obviously working as expected, but for whatever
reason the telco is either not accepting the right format, or not
processing the presented caller ID correctly.
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[asterisk-users] Announcement: Howler-optimised G.729A Solution for Asterisk
[ Optimised G.729A 'Howlet' for Asterisk FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. This is the first in a series of products dubbed 'Howlets' that add highly performant transcoding and signal processing modules to open-source telecoms platforms. The G.729A Howlet ships as a drop-in module for Asterisk or FreeSWITCH, and enables cost-effective transcoding of G.729 and G.729A calls to other codecs. It scales to more than 225 concurrent transcoded calls on a single dual core server, and is licensed on a per concurrent channel basis. You can choose from two licensing models - fixed server perpetual and annual floating, the latter allowing you to 'float' your licensed channels across multiple servers for ultimate flexibility. Our unique floating licenses means you enable G.729A across your infrastructure without the administrative overhead of managing per-server licenses, and at a fraction of the initial cost of fixed-server licenses. Howlets are available for purchase immediately, and start at just £3.99/channnel with all patent holder royalties taken care of. Download your free trial today! http://www.howlertech.com/products/howlets/ -- Enjoy, The Howler Team supp...@howlertech.com Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7099 fax: +44 207 099 7098 http://www.howlertech.com/ Registered in England Wales, Company No. 06285634 ___ -- Bandwidth and Colocation Provided 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] Working chan_mobile/bluez anyone?
Hi all, Before I start with analog GSM gateways I wanted to check if maybe someone actually got a working combination of chan_mobile and bluez. If you do please share specifics like versions, phone, BT chipset, any other relevant info. Thanks, Sasa Bobek ___ -- Bandwidth and Colocation Provided 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 + Jabber
jonas kellens schrieb: I want to use JabberSend in my dialplan, but I saw that my Asterisk does not support Jabber. Also I have nowhere a module res_jabber.so... So I thought I'd rebuild my Asterisk. In menuselect I saw that res_jabber was dependent of 'iksemel' and 'gnutls'. In my yum repositories I can find a gnutls.i386, but what is this iksemel-beast ??? There is info to find via google on the configuration of jabber.conf and the integration of jabber asterisk. But I can not find info on how to build jabber-support with Asterisk. I assume you do this via menuselect. So, what about this 'iksemel' ?? On Debian: aptitude search iksemel You might be on your own on other distros: http://code.google.com/p/iksemel/ Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.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
Re: [asterisk-users] [extensions.conf] Any idea why not working as itshould?
It is a php problem
you need to put a break in your case 2
case 2:break;
case 3:
$mail-AddAddress(User2);
$mail-AddAddress(User1);
break;
Ish
Danny Nicholas wrote:
First of all, this is a PHP problem, not an asterisk one. That being said,
could the empty case 2: be causing the problem? Have you run the php from a
command line to see what happens (php send_call_notification.phpcli 3)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vincent
Sent: Tuesday, June 23, 2009 9:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] [extensions.conf] Any idea why not working as
itshould?
Hello
I noticed a small bug in the way my extensions.conf work:
Users can choose extensions 1-4 or 9 to tell why they're calling, and
I'll send an e-mail to the person(s) to whom is involved. Extension 4
is actually for personal messages for User1, and extension 9 is for
everyone (User1, User2, and User3).
= For some reason, when the caller chooses extension 4, both User1
and User2 get the e-mail, while I expect only User1 to get one:
= /usr/local/etc/asterisk/extensions.conf
//Caller can choose extensions 1, 2, 3, 4 or 9
//Look up software name from extension
exten = _[1-49],1,AGI(convert_app.phpcli|${EXTEN})
exten = _[1-49],n,Set(APPLICATION_NUM=${EXTEN})
//Send e-mail to who is in charge of the application
exten =
_[1-49],n,AGI(send_call_notification.phpcli|${CALLERID(name)}|${CALLERID(num
)}|${APPLICATION_NUM})
exten = _[1-49],n,Hangup()
= /usr/local/share/asterisk/agi-bin/send_call_notification.phpcli
switch($argv[3]) {
//Softare 1
case 1:
$mail-AddAddress(User1);
break;
//Softwares 2,3
case 2:
case 3:
$mail-AddAddress(User2);
$mail-AddAddress(User1);
break;
//Personal msg
//BUG: Why does User2 also receive this e-mail?
case 4:
$mail-AddAddress(User1);
break;
//Any other subject
case 9:
$mail-AddAddress(User3);
$mail-AddAddress(User2);
$mail-AddAddress(User1);
break;
}
= Even when user selects extension 4, e-mail is sent to User1 and
User2!
To: us...@acme.com, us...@acme.com
Subject: [Personal msg for User1] Call from John Doe (555-1234)
===
Any idea why this is?
Thanks for any hint.
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--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] Message Waiting Indication Astersk and kamailio
In the sip conf for the extension make sure you populate the mailbox option and put in mailbox numner@context Ish DHAVAL INDRODIYA wrote: hi all, I have Asterisk 1.6.0.5 Installed and kamailio 1.0.5 version installed when i leave voicemail On Asterisk i need MWI Indication on kamailio extension there are some methods i tried but still cant get success All other feature are working fine also try voip-info.org http://voip-info.org methods can anybody suggest me for different method and have some different setting on SIP . any help appreciated regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided 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] Minimizing downtime during updates
- Original Message - From: Dave Fullerton dfullertaster...@shorelinecontainer.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 23, 2009 8:39 AM Subject: Re: [asterisk-users] Minimizing downtime during updates Karl Fife wrote: I was about to ask this question when I figured out the answer by combing through the makefile. I am posting this anyway because I think it's good to know, and I didn't find any threads that speak to it when I searched the list history. My Question was: When updating Asterisk, the sound tarballs for the selected codecs are not retreived until running make install. This adds unnecessarily to the downtime when updating versions because Asterisk has to be stopped while running make install. I wanted a simple way to pre-fetch these files to a local repository to speed up the actual install routine, instead of slowing it by the arbitrary duration of the fetch/download process which robs valuable NINES from uptime :-) I discovered that after running make, you can run 'make sounds' before shutting down the service. This cuts all of the download time from the install process minimizing service downtime to a fraction of what it would othewise be. You can also just grab and un-tar the sound files by hand from: http://downloads.asterisk.org/pub/telephony/sounds/ On a side note, why does the sounds directory not display in the directory listing when looking at http://downloads.digium.com/pub/telephony/ ? -Dave Karl Fife wrote: Dave Fullerton dfullertaster...@shorelinecontainer.com wrote: You can also just grab and un-tar the sound files by hand from: http://downloads.asterisk.org/pub/telephony/sounds/ Good point because the MOH files would need to be bulled down manually if you want to minimize downtime AND you choose to offer wideband MOH tracks to calling parties (and perhaps other native codecs). It is my observation that make sounds target does not fetch those MOH tracks (as I would have expected), rather they are only fetched during 'make install', (increasing downtime). Does anyone know if there is in fact a distinct target in the makefile that pulls these down, and if not, why they're not pulled down as a matter of course with make sounds if specified in makefile.makeopts. -Karl The trigger (I believe) is when you select the sound packages in menuselect. By default the GSM core sound files and WAV music on hold are included in the asterisk tar file. If you want to pre-download music files then wget them into the /usr/src/asterisk-1.x.x/sounds directory prior to running make. By skimming through the make file in that directory it looks like it tests for their existence prior to downloading them. Make sure you download the version appropriate to that versions of asterisk (you'll have to look in the sounds/Makefile at CORE_SOUNDS_VERSION and EXTRA_SOUNDS_VERSION) and not the -current tar balls. -Dave ___ -- Bandwidth and Colocation Provided 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] Announcement: Howler-optimised G.729A Solution for Asterisk
Jay Fenton wrote: [ Optimised G.729A 'Howlet' for Asterisk FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. Please do not post advertisements for commercial products on this *non-commercial* mailing list. -- 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] Sangoma A200 and bristuff how install
Hi, I want to install an Sangoma A200 together with an BRI card. I would like to use Asterisk 1.4 Are there any howto or tips? First compile bristuff and after compile wanpipe? 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] Announcement: Howler-optimised G.729A Solution for Asterisk
Jay Fenton wrote: [ Optimised G.729A 'Howlet' for Asterisk FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. Why would someone buy it instead of Digium g729 codec? Senad ___ -- Bandwidth and Colocation Provided 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] Announcement: Howler-optimised G.729A Solution for Asterisk
2009/6/24 Senad Jordanovic se...@bicom.us Jay Fenton wrote: [ Optimised G.729A 'Howlet' for Asterisk FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. Why would someone buy it instead of Digium g729 codec? Concurrence is good. And the floating model across many server is interesting idea. ___ -- Bandwidth and Colocation Provided 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] DAHDI Linux 2.2.0 and Tools 2.2.0 Release Announcement
The Asterisk Development Team is pleased to announce the release of DAHDI Linux 2.2.0 and DAHDI Tools 2.2.0. Both releases are available for immediate download at http://downloads.asterisk.org/pub/telephony In addition to various bug fixes, these releases include: * Support for new Xorcom Astribanks with the TwinStar[tm] option. * Improved hardware echo canceler performance for Digium VPMADT032. * Improved fax tone detection and echo canceler / fax handling. * Improved timing accuracy of dahdi_dummy, including when running in virtual environments. * New buffering policy (DAHDI_POLICY_HALF_FULL) which can help faxing performance. * Fixes for Dahdi-perl for non-Xorcom hardware. * BRI Astribank modules no longer need the bri_dchan patch. * Explicit ordering of Astribanks for multi-Astribanks setups. Please report issues found in these releases on http://issues.asterisk.org/. For a full list of the changes in these releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/dahdi-linux/ChangeLog-2.2.0 http://downloads.asterisk.org/pub/telephony/dahdi-tools/ChangeLog-2.2.0 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided 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] Announcement: Howler-optimised G.729A Solution for Asterisk
On Wed, 24 Jun 2009 15:11:42 + (UTC), Jeff LaCoursiere wrote: On Wed, 24 Jun 2009, Grygoriy Dobrovolskyy wrote: 2009/6/24 Senad Jordanovic se...@bicom.us Jay Fenton wrote: [ Optimised G.729A 'Howlet' for Asterisk FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. Why would someone buy it instead of Digium g729 codec? Concurrence is good. And the floating model across many server is interesting idea. I have a question in to them about how that floating licensing works, though. Does that mean that with every call a license check must be made? I don't see how it would work otherwise, and that means my whole business - every call - is dependant on their license server being up and reachable. I also don't think that the slight added convenience is then worth the recurring cost annually. The price of the license is comparable to Digium in US dollars. So the only advantage I really see is the optimization claims - you might be able to squeeze more calls into one box. Would love to hear of any real world experiences, though I guess we will have to wait a bit for that ;) Bear in mind that Digium's present licensing scheme is hardware dependent. That has proven a problem for people wanting to run G.729 on Asterisk in VMs or in EC2. The new lisencing scheme alone has merit. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided 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] Announcement: Howler-optimised G.729A Solution for Asterisk
On Wed, 24 Jun 2009, Grygoriy Dobrovolskyy wrote: 2009/6/24 Senad Jordanovic se...@bicom.us Jay Fenton wrote: [ Optimised G.729A 'Howlet' for Asterisk FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. Why would someone buy it instead of Digium g729 codec? Concurrence is good. And the floating model across many server is interesting idea. I have a question in to them about how that floating licensing works, though. Does that mean that with every call a license check must be made? I don't see how it would work otherwise, and that means my whole business - every call - is dependant on their license server being up and reachable. I also don't think that the slight added convenience is then worth the recurring cost annually. The price of the license is comparable to Digium in US dollars. So the only advantage I really see is the optimization claims - you might be able to squeeze more calls into one box. Would love to hear of any real world experiences, though I guess we will have to wait a bit for that ;) 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] Announcement: Howler-optimised G.729A Solution for Asterisk
Jeff LaCoursiere schrieb: I have a question in to them about how that floating licensing works, though. Does that mean that with every call a license check must be made? I don't see how it would work otherwise, and that means my whole business - every call - is dependant on their license server being up and reachable. I guess that you run your own license server and your machines check the availability of one of your licenses there. At least thats how some other companies for e.g. TTS licenses do it. I also don't think that the slight added convenience is then worth the recurring cost annually. The price of the license is comparable to Digium in US dollars. If you are running a couple of servers and you don't know where your G729 calls will arrive then it makes sense to me. If you run G729 only and have licenses for every line in your system then it obviously makes no sense. But if you have for example 1200 lines and 10% G729 users you never know if they are spread over all your server or all arrive on one machine. Chris ___ -- Bandwidth and Colocation Provided 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] Redundant Connectivity
On Wed, Jun 17, 2009 at 7:10 PM, Marshall Hendersonmarshall...@gmail.com wrote: architecture, etc. On a brand new dual or quad core xeon type system(quite likely multiple physical CPUs, each with multiple cores), And finally, are there any hard or soft limits to be concerned about in regards to the number of simultaneous calls a system can handle? As mentioned, the server function will be purely routing, no other services available. Can each server handle 500 simultaneous calls? More? You don't mention anything about codec for SIP, and that changes the overhead per call. I've done 500 calls on similar gear without breaking a sweat. You should have the same result. I have no clue about your questions with regard to IAX. I'm planing to use Asterisk 1.4.x for this project as it's stable and works very nicely in my existing systems. 1.6.x seems to be a bit too bleeding edge... If there are specific examples why 1.6.x would be a better choice, I'm all ears. Or, is 1.2.x or 1.0.x the way to go? :-) there have been a series of security fixes, so if you go 1.4.x make sure you are going with recent revisions or mitigating the risk of using an old version. You can always read the change log for the 1.6.X versions to find out what you're missing by living with old versions of asterisk. Mostly you're missing changes to underlying performance enhancements, and 'new' features that many of us have been using for a year plus. Some people will say that 1.4 is too bleeding edge. You need to burn in any solution you choose if you want to be satisfied that the result scales appropriately and reliably. Callfiles, a while loop, and logging come in handy there. ___ -- Bandwidth and Colocation Provided 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 can I get Better natural Voice in Festival
On Thu, Jun 18, 2009 at 2:16 AM, DHAVAL INDRODIYAdhaval.it01...@gmail.com wrote: hello All I am using festival as an application but it default voice is not good to hear anybody have solution about better voice in Festival I'm of the opinion that festival is: a) pretty good b) better than it used to be if you use the newer algorithm whose name I'm forgetting. If you don't like the output of the free TTS, you can try commercial products instead. Some people say you get what you pay for. You can read the archives for opinions about various commercial TTS products. And you can always hire professional voice talent if you are recording prompts. ___ -- Bandwidth and Colocation Provided 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] Announcement: Howler-optimised G.729A Solution for Asterisk
On Wed, Jun 24, 2009 at 10:58 AM, Grygoriy Dobrovolskyy megaho...@gmail.com wrote: 2009/6/24 Senad Jordanovic se...@bicom.us Jay Fenton wrote: [ Optimised G.729A 'Howlet' for Asterisk FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. Why would someone buy it instead of Digium g729 codec? Concurrence is good. And the floating model across many server is interesting idea. Similar to Trixter's idea way back when but his was more of a community share. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided 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 can I get Better natural Voice in Festival
On Wed, 24 Jun 2009, David Backeberg wrote: On Thu, Jun 18, 2009 at 2:16 AM, DHAVAL INDRODIYAdhaval.it01...@gmail.com wrote: I am using festival as an application but it default voice is not good to hear anybody have solution about better voice in Festival If you don't like the output of the free TTS, you can try commercial products instead. Some people say you get what you pay for. You can read the archives for opinions about various commercial TTS products. And you can always hire professional voice talent if you are recording prompts. I like Cepstral with the Allison font. (The Voice of Asterisk.) Sometimes a bit obvious, sometimes indistinguishable from her recorded prompts. If I took the time to learn their markup language, I think I'd be even happier. I think it was only $30 with a Nerd Vittles coupon code, but it's been a long time... 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] GUI for Asterisk
I wonder if there is a GUI that does not change the underlying hand-made configuration ?! What I'm looking for actually is a GUI for adding a new SIP-client + voicemail, so that a company does not have to call me when they hired a new employee. I don't want a GUI that over-writes my hand-made SIP-configuration, and my hand-made dialplan. 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
Re: [asterisk-users] GUI for Asterisk
On Wed, Jun 24, 2009 at 3:20 PM, jonas kellens jonas.kell...@telenet.bewrote: I wonder if there is a GUI that does not change the underlying hand-made configuration ?! What I'm looking for actually is a GUI for adding a new SIP-client + voicemail, so that a company does not have to call me when they hired a new employee. I don't want a GUI that over-writes my hand-made SIP-configuration, and my hand-made dialplan. Jonas. FreePBX and I suspect most others have includes to custom files. Put your static stuff in the custom file. Example for FreePBX extensions_custom.conf. That will survive a reload or even an upgrade. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided 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] GUI for Asterisk
Change writeprotect = no to writeprotect = yes. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Wednesday, June 24, 2009 2:21 PM To: Asterisk Mailing Subject: [asterisk-users] GUI for Asterisk I wonder if there is a GUI that does not change the underlying hand-made configuration ?! What I'm looking for actually is a GUI for adding a new SIP-client + voicemail, so that a company does not have to call me when they hired a new employee. I don't want a GUI that over-writes my hand-made SIP-configuration, and my hand-made dialplan. 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] PHP AGI Not Working and Odd Behavior
Hi, I'm running asterisk 1.4.22 on a debian server. I have php5 installed and it works correctly command line. When trying to run a php script via AGI, I get messages such as: GI Tx I AGI Rx #!/usr/bin/php5 -q AGI Tx 510 Invalid or unknown command The scripts are completely executable and owned by asterisk -rwxr-xr-x 1 asterisk asterisk Googling is not helping much, and php seems installed perfectly. Other servers running the same AGIs have no such problem. I also have noticed odd behavior. When I edit an AGI, the changes aren't always showing up in the running of the script via asterisk. For example, I tried editing the bash command to read #!/usr/bin/php -q, and got the same response on my agi debug. I know for a fact it's running the script I've edited: Launched AGI Script /var/lib/asterisk/agi-bin/myscript.php and it's not in some other directory. Any input: obvious or not is requested...a few people here are stumped! Thank you! Leah Newmark VoIP Programmer Capalon Communications ___ -- Bandwidth and Colocation Provided 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] video call doesn work
i am trying to make a video call on asterisk 1.6 , my configuration is an - asterisk 1.6 on Centos on virtual machine VmWare - Xlite softphone one windows xp (the Host operating system) - X-lite client on another windows XP (the Guest operating system ) i put the paramtervideosupport=yes under the general section in sip.conf i allowed the video codecs for each client in sip.conf for the clients 3500 and 3501 i installed 2 web cams one for each client , and in the X-lite video side-window each cam operate well on its corresponding X-lite client in the down part, and when i start a call from 3500 to 3501 and the call established and i press the send video button on both clients , but the video stream is not sent to any of the 2 clients what's wrong? am i missing something? or does the VmWare enviroment cause the problem and i need 2 seperate physical machines Gres___ -- Bandwidth and Colocation Provided 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] video call doesn work
Make sure the video codecs in the xlite setup are also in sip.conf (allow=ulaw,alaw,gsm,h263) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of gmail Sent: Thursday, June 25, 2009 12:57 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] video call doesn work i am trying to make a video call on asterisk 1.6 , my configuration is an - asterisk 1.6 on Centos on virtual machine VmWare - Xlite softphone one windows xp (the Host operating system) - X-lite client on another windows XP (the Guest operating system ) i put the paramtervideosupport=yes under the general section in sip.conf i allowed the video codecs for each client in sip.conf for the clients 3500 and 3501 i installed 2 web cams one for each client , and in the X-lite video side-window each cam operate well on its corresponding X-lite client in the down part, and when i start a call from 3500 to 3501 and the call established and i press the send video button on both clients , but the video stream is not sent to any of the 2 clients what's wrong? am i missing something? or does the VmWare enviroment cause the problem and i need 2 seperate physical machines Gres ___ -- Bandwidth and Colocation Provided 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] PHP AGI Not Working and Odd Behavior
On Wed, Jun 24, 2009 at 3:43 PM, Leah Newmarklnewm...@capalon.com wrote: Hi, I'm running asterisk 1.4.22 on a debian server. I have php5 installed and it works correctly command line. When trying to run a php script via AGI, I get messages such as: GI Tx I AGI Rx #!/usr/bin/php5 -q AGI Tx 510 Invalid or unknown command The scripts are completely executable and owned by asterisk -rwxr-xr-x 1 asterisk asterisk Googling is not helping much, and php seems installed perfectly. Other servers running the same AGIs have no such problem. I also have noticed odd behavior. When I edit an AGI, the changes aren't always showing up in the running of the script via asterisk. For example, I tried editing the bash command to read #!/usr/bin/php -q, and got the same response on my agi debug. I know for a fact it's running the script I've edited: Launched AGI Script /var/lib/asterisk/agi-bin/myscript.php and it's not in some other directory. Keep in mind that if you change your dialplan to call a different script you will need to cli dialplan reload Other than that, I'm not sure that it's legal to put an argument into a shbang, as in your -q when launching php. It's also possible you've somehow locked down php or directories way too much. The proper test is to: bash$ sudo -u asterisk /path/to/script.php ___ -- Bandwidth and Colocation Provided 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] PHP AGI Not Working and Odd Behavior
Thanks. I didn't change anything in my dialplan. I am aware of reloading configuration :) My AGIs are copied from a working asterisk install -- the shebang argument is how I've always done it. Either way, I have tried it without the -q as well, and that also didn't succeed. I just tried your test and it worked fine to run it. As I said, I know the server is reading the file I've been editing. I see it on the monitor. It's definitely opening the file to return that message... Any other ideas? On Wed, Jun 24, 2009 at 3:43 PM, Leah Newmarklnewmark at capalon.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / Hi, // // I'm running asterisk 1.4.22 on a debian server. // I have php5 installed and it works correctly command line. // When trying to run a php script via AGI, I get messages such as: // GI Tx I // AGI Rx #!/usr/bin/php5 -q // AGI Tx 510 Invalid or unknown command // // The scripts are completely executable and owned by asterisk // -rwxr-xr-x 1 asterisk asterisk // // Googling is not helping much, and php seems installed perfectly. Other // servers running the same AGIs have no such problem. // // I also have noticed odd behavior. When I edit an AGI, the changes aren't // always showing up in the running of the script via asterisk. // For example, I tried editing the bash command to read #!/usr/bin/php -q, // and got the same response on my agi debug. // I know for a fact it's running the script I've edited: // Launched AGI Script /var/lib/asterisk/agi-bin/myscript.php // and it's not in some other directory. / Keep in mind that if you change your dialplan to call a different script you will need to cli dialplan reload Other than that, I'm not sure that it's legal to put an argument into a shbang, as in your -q when launching php. It's also possible you've somehow locked down php or directories way too much. The proper test is to: bash$ sudo -u asterisk /path/to/script.php ___ -- Bandwidth and Colocation Provided 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] video call doesn work
i already did that - Original Message - From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, June 24, 2009 1:08 PM Subject: Re: [asterisk-users] video call doesn work Make sure the video codecs in the xlite setup are also in sip.conf (allow=ulaw,alaw,gsm,h263) -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of gmail Sent: Thursday, June 25, 2009 12:57 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] video call doesn work i am trying to make a video call on asterisk 1.6 , my configuration is an - asterisk 1.6 on Centos on virtual machine VmWare - Xlite softphone one windows xp (the Host operating system) - X-lite client on another windows XP (the Guest operating system ) i put the paramtervideosupport=yes under the general section in sip.conf i allowed the video codecs for each client in sip.conf for the clients 3500 and 3501 i installed 2 web cams one for each client , and in the X-lite video side-window each cam operate well on its corresponding X-lite client in the down part, and when i start a call from 3500 to 3501 and the call established and i press the send video button on both clients , but the video stream is not sent to any of the 2 clients what's wrong? am i missing something? or does the VmWare enviroment cause the problem and i need 2 seperate physical machines Gres -- ___ -- Bandwidth and Colocation Provided 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] GUI for Asterisk
On Wed, Jun 24, 2009 at 09:20:44PM +0200, jonas kellens wrote: I wonder if there is a GUI that does not change the underlying hand-made configuration ?! What I'm looking for actually is a GUI for adding a new SIP-client + voicemail, so that a company does not have to call me when they hired a new employee. I don't want a GUI that over-writes my hand-made SIP-configuration, and my hand-made dialplan. You're looking at it the wrong way. Figure out where the GUI generates / updates the configuration and make sure it gets things right. Either you write configuration manually or the GUI writes them. Don't try mixing both too badly. -- 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
[asterisk-users] Determining picked up line from multiple line ring
Hi all, I've looked at the various variables but can't seem to find a way to determine which line was picked up in a multi-line ring. For example, in this excerpt from my asterisk logging: -- Executing [5558280...@inbound:52] Dial(SIP/proxy3-05ac9180, SIP/1555...@proxy1SIP/1555...@proxy1|18|r) in new stack -- Called 1555...@proxy1 -- Called 1555...@proxy1 -- SIP/proxy1-05af5ca0 is making progress passing it to SIP/proxy3-05ac9180 -- SIP/proxy1-05acaae0 is making progress passing it to SIP/proxy3-05ac9180 -- SIP/proxy1-05acaae0 answered SIP/proxy3-05ac9180 -- Packet2Packet bridging SIP/proxy3-05ac9180 and SIP/proxy1-05acaae0 When someone dials in to 555828, I call two phone numbers, 1555111 and 1555222 simultaneously. The logging shows when one of those numbers is picked up, but I don't know which one. I'd like to be able to determine which phone number was picked up. How do I do that? Is there a variable somewhere I can tap in real time? The CDRs don't show which number was picked up either. Thanks! Enlai ___ -- Bandwidth and Colocation Provided 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] video call doesn work
On Thu, 2009-06-25 at 10:56 -0700, gmail wrote: i am trying to make a video call on asterisk 1.6 Video support in Asterisk 1.6.0 and later appears to be broken. I have a hackish patch that makes *some* calls work, but it's not an elegant fix. See https://issues.asterisk.org/view.php?id=15121 for more details. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determining picked up line from multiple line ring
${CHANNEL} or ${DNID} should do the trick.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
asterisk-us...@enlai.net
Sent: Wednesday, June 24, 2009 3:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Determining picked up line from multiple line ring
Hi all,
I've looked at the various variables but can't seem to find a way to
determine which line was picked up in a multi-line ring.
For example, in this excerpt from my asterisk logging:
-- Executing [5558280...@inbound:52] Dial(SIP/proxy3-05ac9180,
SIP/1555...@proxy1SIP/1555...@proxy1|18|r) in new stack
-- Called 1555...@proxy1
-- Called 1555...@proxy1
-- SIP/proxy1-05af5ca0 is making progress passing it to
SIP/proxy3-05ac9180
-- SIP/proxy1-05acaae0 is making progress passing it to
SIP/proxy3-05ac9180
-- SIP/proxy1-05acaae0 answered SIP/proxy3-05ac9180
-- Packet2Packet bridging SIP/proxy3-05ac9180 and
SIP/proxy1-05acaae0
When someone dials in to 555828, I call two phone numbers,
1555111 and 1555222 simultaneously.
The logging shows when one of those numbers is picked up, but I don't
know which one. I'd like to be able to determine which phone number was
picked up. How do I do that? Is there a variable somewhere I can tap in
real time? The CDRs don't show which number was picked up either.
Thanks!
Enlai
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Re: [asterisk-users] GUI for Asterisk
On Wed, Jun 24, 2009 at 4:39 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, Jun 24, 2009 at 09:20:44PM +0200, jonas kellens wrote: I wonder if there is a GUI that does not change the underlying hand-made configuration ?! What I'm looking for actually is a GUI for adding a new SIP-client + voicemail, so that a company does not have to call me when they hired a new employee. I don't want a GUI that over-writes my hand-made SIP-configuration, and my hand-made dialplan. You're looking at it the wrong way. Figure out where the GUI generates / updates the configuration and make sure it gets things right. Either you write configuration manually or the GUI writes them. Don't try mixing both too badly. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir In FreePBX there are whatever_custom.conf files that are not touched when changes are made in the GUI. Only downside I see, and I think it applies to included files as it does to included contexts is that pattern matching will occur in the first context (or file) even if there is another pattern match in the second. Not sure how that could hurt, but I am sure someone would find a reason/time where it would. With FreePBX and others, the whatever_additional.conf file is what gets overwritten when changes are made in the gui and applied. If you want to get fancy, you can jump into the FreePBX DB and set the values yourself so that they are written by the GUI when applied. Probably the cleanest way. Another semi static option is to edit the whatever.conf file. Out of these three options, only the editing of whatever_custom.conf will survive certain upgrades. Then chmod that bad boy so it can only be read. I think most systems work similarly. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided 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] Removing line 2 from CISCO 7940g
The phone caches the configuration... To remove it update the config like so: line2_name:UNPROVISIONED line2_authname:UNPROVISIONED line2_password:UNPROVISIONED line2_shortname: UNPROVISIONED line2_displayname: UNPROVISIONED For each line that you don't want anymore. So on a 7960 you would have to do this for lines 2-6. The line will then disappear from the phone. -Jonathan On Wed, Jun 24, 2009 at 2:11 PM, Mike asterisk-us...@norgie.net wrote: Folks, I have CISCO 7940g phone. I have in the past configured the phone with two lines. Having found the 2nd line wasn't much use, I want to remove it from the config. I have taken it out of the SIP config file that is TFTPd to the phone but it is still showing on the phone and it is still trying to log into Asterisk with that account. I have tried removing the config line and blanking out the options but it still persists. Does anyoen know how to get rid of the thing? Mike. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkpCloMACgkQmUrfmTU1ohWLzwCg39To92tTSB+6j8TkkJ4QTO+S 1cAAn3a7FvqwKu4Id/LV44JiO8rmR4m/ =Dpe0 -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
Re: [asterisk-users] PHP AGI Not Working and Odd Behavior
Looking at my man php5 q is not a valid option. That may be just on Suse. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. Rodríguez Sent: Wednesday, June 24, 2009 4:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PHP AGI Not Working and Odd Behavior Try running your script with /usr/bin/php5 script.php to test it Or changing #!/usr/bin/php5 -q to #!/usr/bin/php -q Leah Newmark wrote: Thanks. I didn't change anything in my dialplan. I am aware of reloading configuration :) My AGIs are copied from a working asterisk install -- the shebang argument is how I've always done it. Either way, I have tried it without the -q as well, and that also didn't succeed. I just tried your test and it worked fine to run it. As I said, I know the server is reading the file I've been editing. I see it on the monitor. It's definitely opening the file to return that message... Any other ideas? On Wed, Jun 24, 2009 at 3:43 PM, Leah Newmarklnewmark at capalon.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / Hi, // // I'm running asterisk 1.4.22 on a debian server. // I have php5 installed and it works correctly command line. // When trying to run a php script via AGI, I get messages such as: // GI Tx I // AGI Rx #!/usr/bin/php5 -q // AGI Tx 510 Invalid or unknown command // // The scripts are completely executable and owned by asterisk // -rwxr-xr-x 1 asterisk asterisk // // Googling is not helping much, and php seems installed perfectly. Other // servers running the same AGIs have no such problem. // // I also have noticed odd behavior. When I edit an AGI, the changes aren't // always showing up in the running of the script via asterisk. // For example, I tried editing the bash command to read #!/usr/bin/php -q, // and got the same response on my agi debug. // I know for a fact it's running the script I've edited: // Launched AGI Script /var/lib/asterisk/agi-bin/myscript.php // and it's not in some other directory. / Keep in mind that if you change your dialplan to call a different script you will need to cli dialplan reload Other than that, I'm not sure that it's legal to put an argument into a shbang, as in your -q when launching php. It's also possible you've somehow locked down php or directories way too much. The proper test is to: bash$ sudo -u asterisk /path/to/script.php ___ -- Bandwidth and Colocation Provided 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] Determining picked up line from multiple line ring
I think I got it. ${DIALEDPEERNUMBER} contains the leg that connected
(just what I need).
FYI I used DumpChan() to get all the available variables and found it.
Thanks!
Enlai
On Wed, 24 Jun 2009 14:19:46 -0700, asterisk-users@lists.digium.com
said:
Thanks Danny.
I tried accessing ${CHANNEL} and ${DNID} or ${CALLERID{dnid)} in the h
(hangup) context, which is invoked after either party hangs up.
However, the ${CHANNEL} contains the original channel the call came in
on and not the outbound channel that connected. The ${CALLERID(dnid)}
contains the caller's phone number and not the one that connected on the
outbound leg.
Any other ideas? Should I put the ${CHANNEL} and ${CALLERID(dnid)}
somewhere else?
Thanks,
Enlai
On Wed, 24 Jun 2009 15:51:18 -0500, Danny Nicholas da...@debsinc.com
said:
${CHANNEL} or ${DNID} should do the trick.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
asterisk-us...@enlai.net
Sent: Wednesday, June 24, 2009 3:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Determining picked up line from multiple line
ring
Hi all,
I've looked at the various variables but can't seem to find a way to
determine which line was picked up in a multi-line ring.
For example, in this excerpt from my asterisk logging:
-- Executing [5558280...@inbound:52] Dial(SIP/proxy3-05ac9180,
SIP/1555...@proxy1SIP/1555...@proxy1|18|r) in new stack
-- Called 1555...@proxy1
-- Called 1555...@proxy1
-- SIP/proxy1-05af5ca0 is making progress passing it to
SIP/proxy3-05ac9180
-- SIP/proxy1-05acaae0 is making progress passing it to
SIP/proxy3-05ac9180
-- SIP/proxy1-05acaae0 answered SIP/proxy3-05ac9180
-- Packet2Packet bridging SIP/proxy3-05ac9180 and
SIP/proxy1-05acaae0
When someone dials in to 555828, I call two phone numbers,
1555111 and 1555222 simultaneously.
The logging shows when one of those numbers is picked up, but I don't
know which one. I'd like to be able to determine which phone number was
picked up. How do I do that? Is there a variable somewhere I can tap in
real time? The CDRs don't show which number was picked up either.
Thanks!
Enlai
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Re: [asterisk-users] GUI for Asterisk
On Wed, Jun 24, 2009 at 05:01:04PM -0400, Steve Totaro wrote: In FreePBX there are whatever_custom.conf files that are not touched when changes are made in the GUI. The _custom file is not touched. But it is merely part of the configuration file. And what if you want the luser to be able to configure things from the GUI? That luser that does not know how to manually fix the _custom.conf file? -- 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] PHP AGI Not Working and Odd Behavior
Try running your script with /usr/bin/php5 script.php to test it Or changing #!/usr/bin/php5 -q to #!/usr/bin/php -q Leah Newmark wrote: Thanks. I didn't change anything in my dialplan. I am aware of reloading configuration :) My AGIs are copied from a working asterisk install -- the shebang argument is how I've always done it. Either way, I have tried it without the -q as well, and that also didn't succeed. I just tried your test and it worked fine to run it. As I said, I know the server is reading the file I've been editing. I see it on the monitor. It's definitely opening the file to return that message... Any other ideas? On Wed, Jun 24, 2009 at 3:43 PM, Leah Newmarklnewmark at capalon.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / Hi, // // I'm running asterisk 1.4.22 on a debian server. // I have php5 installed and it works correctly command line. // When trying to run a php script via AGI, I get messages such as: // GI Tx I // AGI Rx #!/usr/bin/php5 -q // AGI Tx 510 Invalid or unknown command // // The scripts are completely executable and owned by asterisk // -rwxr-xr-x 1 asterisk asterisk // // Googling is not helping much, and php seems installed perfectly. Other // servers running the same AGIs have no such problem. // // I also have noticed odd behavior. When I edit an AGI, the changes aren't // always showing up in the running of the script via asterisk. // For example, I tried editing the bash command to read #!/usr/bin/php -q, // and got the same response on my agi debug. // I know for a fact it's running the script I've edited: // Launched AGI Script /var/lib/asterisk/agi-bin/myscript.php // and it's not in some other directory. / Keep in mind that if you change your dialplan to call a different script you will need to cli dialplan reload Other than that, I'm not sure that it's legal to put an argument into a shbang, as in your -q when launching php. It's also possible you've somehow locked down php or directories way too much. The proper test is to: bash$ sudo -u asterisk /path/to/script.php ___ -- Bandwidth and Colocation Provided 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] Removing line 2 from CISCO 7940g
Folks, I have CISCO 7940g phone. I have in the past configured the phone with two lines. Having found the 2nd line wasn't much use, I want to remove it from the config. I have taken it out of the SIP config file that is TFTPd to the phone but it is still showing on the phone and it is still trying to log into Asterisk with that account. I have tried removing the config line and blanking out the options but it still persists. Does anyoen know how to get rid of the thing? Mike. signature.asc Description: Digital 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
Re: [asterisk-users] GUI for Asterisk
On Wed, Jun 24, 2009 at 5:31 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, Jun 24, 2009 at 05:01:04PM -0400, Steve Totaro wrote: In FreePBX there are whatever_custom.conf files that are not touched when changes are made in the GUI. The _custom file is not touched. But it is merely part of the configuration file. And what if you want the luser to be able to configure things from the GUI? That luser that does not know how to manually fix the _custom.conf file? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir And your point being? Mountain, molehill. Did I not address what the OP was asking They can add SIP clients, VM, or whatever changes when a company hires a new employee via the GUI. It also does not change the underlying hand-made configurations. sip_custom.conf extensions_custom.conf Done and done. Please re-read to be clear.. *I wonder if there is a GUI that does not change the underlying hand-made configuration ?! What I'm looking for actually is a GUI for adding a new SIP-client + voicemail, so that a company does not have to call me when they hired a new employee. I don't want a GUI that over-writes my hand-made SIP-configuration, and my hand-made dialplan. * *Jonas. * -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided 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 set the II DIgits?
I need to set the II digits for some outgoing calls originating with asterisk, but the documentation seems to show that all the various ANI2 variables are read-only. So how do I set them? (Yes, we have Feature Group D trunks and allowed to set them and regularly do with our C.O. switch. The interface between that switch and asterisk is via ISDN spans.) Thanks... Jim Gottlieb San Diego, California ___ -- Bandwidth and Colocation Provided 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] dahdi-linux-2.2.0 compile problem
I have an i686 cpu and when compiling from source I get this error: touch /usr/src/dahdi-linux-2.2.0/drivers/dahdi/xpp/init_fxo_modes.verified Building modules, stage 2. MODPOST WARNING: could not find /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32. o.cmd for /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o Anyone else seeing this? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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] GUI for Asterisk
On Wed, Jun 24, 2009 at 05:41:34PM -0400, Steve Totaro wrote: On Wed, Jun 24, 2009 at 5:31 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, Jun 24, 2009 at 05:01:04PM -0400, Steve Totaro wrote: In FreePBX there are whatever_custom.conf files that are not touched when changes are made in the GUI. The _custom file is not touched. But it is merely part of the configuration file. And what if you want the luser to be able to configure things from the GUI? That luser that does not know how to manually fix the _custom.conf file? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir And your point being? Mountain, molehill. Did I not address what the OP was asking They can add SIP clients, VM, or whatever changes when a company hires a new employee via the GUI. It also does not change the underlying hand-made configurations. Until you create (through the GUI) an extension or trunk with some strange name :-) -- 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] dahdi-linux-2.2.0 compile problem
On Wed, Jun 24, 2009 at 03:53:18PM -0700, Jim Dickenson wrote: I have an i686 cpu and when compiling from source I get this error: touch /usr/src/dahdi-linux-2.2.0/drivers/dahdi/xpp/init_fxo_modes.verified Building modules, stage 2. MODPOST WARNING: could not find /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32. o.cmd for /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o Anyone else seeing this? My guess: a failed download? A partial download? -- 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] dahdi-linux-2.2.0 compile problem
I download the tar.gz file and expand it, without error. I am not sure how I could not have a complete download. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ From: Tzafrir Cohen tzafrir.co...@xorcom.com Organization: Xorcom* Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 25 Jun 2009 02:07:14 +0300 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] dahdi-linux-2.2.0 compile problem On Wed, Jun 24, 2009 at 03:53:18PM -0700, Jim Dickenson wrote: I have an i686 cpu and when compiling from source I get this error: touch /usr/src/dahdi-linux-2.2.0/drivers/dahdi/xpp/init_fxo_modes.verified Building modules, stage 2. MODPOST WARNING: could not find /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32. o.cmd for /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o Anyone else seeing this? My guess: a failed download? A partial download? -- 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 ___ -- Bandwidth and Colocation Provided 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-linux-2.2.0 compile problem
Jim Dickenson wrote: Building modules, stage 2. MODPOST WARNING: could not find /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32. o.cmd for /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o Anyone else seeing this? You can safely ignore these warnings. The kernel build system uses the .cmd files in order to calculate dependencies and module version information as part of the build. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
