Re: [asterisk-users] Is it possible that variables returned from AGI take a moment to "stick"?
I don't think so any such method to return variable from AGI. But simple solution is set variable in AGI and then you can get back after AGI call in dialplan and these variable will be available until call finished. --- Virendra Bhati +91-9718500594 +91-9250078532 Sr. Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) On Fri, Oct 21, 2016 at 6:06 PM, Jonathan H <lardconce...@gmail.com> wrote: > I thought dialplan flow was that (normal!) agi was called, it did its > thing (which include returning some dialplan variables/lists), and > then when agi finished it returned to the dialplan which then reliably > carried the product of agi. > > But I'm calling agi, scanning a path in python, and then finding that > unless I call a 1 second wait in the dialplan AFTER the agi, sometimes > the variable is empty, even though agi debug shows it was sent. > > Any tests I can do, or is this to be expected? > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk ARA with Multi tenant solution
Hi team, I had implementation complete customized IPPBX solution with the help on Asterisk , ARA and a2billing for billing purpose. Now only issue I come is if a customer A and B want to used similar extension rang then it's only possible with adding account-code like 100e12345 and 100e67890. But in GUI I will manage display and other features but issue come when customers want to register with Asterisk service. So customers will try 100 not 100e12345, or 100e67890. So my question is can we by pass this or any other alternative solution by which we may got solution? Any clue will be appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.12 segfault
we are also facing an issue in Asterisk 11.4.0 as well. What is the route case of this issue is anyone know ? On Thu, Aug 28, 2014 at 5:32 PM, Grant Bagdasarian g...@cm.nl wrote: Hello, Yes, we use FreeSWITCH primarily for our main platform. Works like a charm! But we also have some applications running on Asterisk (older versions) which can’t be upgraded without careful planning and testing. Anyways, thanks for the response! Grant *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Vik Killa *Sent:* Thursday, August 28, 2014 1:56 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk 1.6.2.12 segfault Grant, Perhaps it's time to upgrade? I used to see tons of unexplained segfaults in 1.6.X, haven't seen any in 1.8.22.0 (I'm afraid to upgrade since I finally found a stable version) You should, also, have you heard of FreeSWITCH? IMO much more stable PBX software. Thanks On Thu, Aug 28, 2014 at 5:45 AM, Grant Bagdasarian g...@cm.nl wrote: Hello, Could someone explain to me what this means? asterisk[30269]: segfault at 0008 rip 2aaac8b388f2 rsp 40a75910 error 4 Also, would this segfault crash the whole Asterisk process or will Asterisk continue to run? Is it possible this would affect/disconnect “SOME” DAHDI channels, but not all? At this point, upgrading is not an option, even though I agree we should. Regards, Grant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Virendra Bhati +91-9718500594 +91-9250078532 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn] http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AppKonference 2.5
Good Paul, I used Konference a lot very nice apps, but will this work with asterisk latest version or not ? I used asterisk 1.4,1.8 but didn't work on 11... On Mon, Dec 16, 2013 at 10:21 PM, Paul Albrecht palbre...@glccom.comwrote: Hi, I have released AppKonference 2.5 today. This release fixes a bug that can cause audio problems when conference frame caching is enabled. It also fixes the spy feature so that more than one spyer can spy on a channel at the same time. If more than one spyer is unmuted, their audio is mixed and whispered to the spyee. -- Paul Albrecht -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Virendra Bhati +91-9718500594 +91-9250078532 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is this big of new modification in Asterisk Events Objects values ?
Hi Team, Thanks for your great job an Asterisk new features developments. I installed asterisk-12 Beta and found some changes as well which i notice to put in-front of your knowledge, don't know that bug of new modification into objects or old version (asterisk-11) mistake corrected that time, anyway *Asterisk-12:* Array ( [Event] = ConfbridgeMute [Privilege] = call,all [Conference] = 42 [BridgeUniqueid] = 9f2ae5df-0749-4494-b8b7-12eb50dc765d [BridgeType] = base [BridgeTechnology] = softmix [BridgeNumChannels] = 2 [Channel] = SIP/5000-0006 [ChannelState] = 6 [ChannelStateDesc] = Up *[CallerIDNum] = 5000* [CallerIDName] = 5000 [ConnectedLineNum] = unknown [ConnectedLineName] = unknown [AccountCode] = [Context] = from-sip [Exten] = 1234 [Priority] = 3 [Uniqueid] = 1382599433.22 ) Please check the BOLD section. earlier is was *[CallerIDnum] * *So 'n' is now 'N' * -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 Asterisk Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is this big of new modification in Asterisk Events Objects values ?
For me that's matter bcoz i was working with events programming and face issue then I notice ,.. On Fri, Oct 25, 2013 at 5:16 PM, jg webaccou...@jgoettgens.de wrote: Does it matter? I thought keys are case insensitive. jg -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 Asterisk engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-12 issue after successful installation
Hi Team,
After suggested links and patch ,
I installed all and then start asterisk and that start working.
Thanks for suggestion..
On Wed, Oct 23, 2013 at 3:55 AM, Matthew Jordan mjor...@digium.com wrote:
On Mon, Oct 21, 2013 at 7:59 AM, A J Stiles asterisk_l...@earthshod.co.uk
wrote:
On Monday 21 October 2013, virendra bhati wrote:
Hi Team,
I have installed asterisk-12 Beta but when I try to asterisk start then
get
below issue.
*[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r
asterisk: error while loading shared libraries: libjansson.so.4: cannot
open shared object file: No such file or directory
[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]#*
Did you build it yourself from Source Code, or did you install someone
else's
pre-compiled package? If the latter, the packager may have omitted a
dependency. It happens from time to time. You probably need to install
a -
dev or -devel package (what distro are you running?)
What do you get for
# ldd /usr/sbin/asterisk
?
Hello -
libjansson is now a required library. Please see the build system changes
in the UPGRADE notes [1] or on the wiki [2]. Note: if you have not yet read
the upgrade notes and the list of changes, please do so before installing
and running Asterisk 12. Please :-)
Note that if your distro doesn't have a package of libjansson (or, more
accurately, libjansson-dev{el}), you can download a source tarball and
install it [3]. The install_prereq script [4] should also take care of it
for you.
[1] http://svn.asterisk.org/svn/asterisk/branches/12/UPGRADE.txt
[2] https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12
[3] http://www.digip.org/jansson/
[4]
http://svn.asterisk.org/svn/asterisk/branches/12/contrib/scripts/install_prereq
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
--
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Thanks and regards
Virendra Bhati
+91-9718500594
+91-9250078532
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755
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Re: [asterisk-users] Asterisk 12 issue
Thank you, My issue was resolved by provided information On Thu, Oct 24, 2013 at 5:49 AM, Sylvain Boily sbo...@proformatique.comwrote: Hello, Le 2013-10-21 08:31, virendra bhati a écrit : Hi Team, I have installed asterisk-12 Beta but when I try to asterisk start then get below issue. *[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r asterisk: error while loading shared libraries: libjansson.so.4: cannot open shared object file: No such file or directory * - Asterisk now depends on libjansson, libuuid and optionally (but recommended) libxslt and uriparser. information from https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12 Sylvain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-12 issue after successful installation
Yes I installed manually from tar file of jansson On Wed, Oct 23, 2013 at 8:44 AM, Warren Selby wcse...@selbytech.com wrote: On Mon, Oct 21, 2013 at 7:26 AM, virendra bhati virbh...@gmail.comwrote: Hi Team, I have installed asterisk-12 Beta but when I try to asterisk start then get below issue. *[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r asterisk: error while loading shared libraries: libjansson.so.4: cannot open shared object file: No such file or directory [root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]#* So, as a specific answer to the original question, the proper resolution to this issue, assuming you manually installed libjansson, is the following, pulled from the install_prereq scripts: echo /usr/local/lib /etc/ld.so.conf.d/usr_local.conf /sbin/ldconfig This worked for me on a fresh CentOS 6.4 installation where I didn't use the install_prereq script, and thus was having your same issue. Hope this helps someone in the future! -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-12 issue after successful installation
Hi , Below is the details of your provided linux command [root@cs-gb-pwr-1-04 ~]# ldd /usr/sbin/asterisk * linux-vdso.so.1 = (0x7fffd29c9000) libasteriskssl.so.1 = /usr/lib64/libasteriskssl.so.1 (0x7ffa226ea000) libc.so.6 = /lib64/libc.so.6 (0x003456c0) libxml2.so.2 = /usr/lib64/libxml2.so.2 (0x003459c0) libz.so.1 = /lib64/libz.so.1 (0x003457c0) libm.so.6 = /lib64/libm.so.6 (0x00345780) libsqlite3.so.0 = /usr/lib64/libsqlite3.so.0 (0x00345880) libssl.so.10 = /usr/lib64/libssl.so.10 (0x00345bc0) libcrypto.so.10 = /usr/lib64/libcrypto.so.10 (0x00345a00) libjansson.so.4 = not found libuuid.so.1 = /lib64/libuuid.so.1 (0x7ffa224e3000) libcrypt.so.1 = /lib64/libcrypt.so.1 (0x00345940) libdl.so.2 = /lib64/libdl.so.2 (0x00345700) libpthread.so.0 = /lib64/libpthread.so.0 (0x00345740) libtinfo.so.5 = /lib64/libtinfo.so.5 (0x00345ac0) libresolv.so.2 = /lib64/libresolv.so.2 (0x003458c0) /lib64/ld-linux-x86-64.so.2 (0x00345680) libgssapi_krb5.so.2 = /lib64/libgssapi_krb5.so.2 (0x00345a80) libkrb5.so.3 = /lib64/libkrb5.so.3 (0x00345b80) libcom_err.so.2 = /lib64/libcom_err.so.2 (0x00345980) libk5crypto.so.3 = /lib64/libk5crypto.so.3 (0x00345b00) libfreebl3.so = /lib64/libfreebl3.so (0x00345900) libkrb5support.so.0 = /lib64/libkrb5support.so.0 (0x00345a40) libkeyutils.so.1 = /lib64/libkeyutils.so.1 (0x00345b40) libselinux.so.1 = /lib64/libselinux.so.1 (0x00345800)* [root@cs-gb-pwr-1-04 ~]# On Mon, Oct 21, 2013 at 6:29 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Monday 21 October 2013, virendra bhati wrote: Hi Team, I have installed asterisk-12 Beta but when I try to asterisk start then get below issue. *[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r asterisk: error while loading shared libraries: libjansson.so.4: cannot open shared object file: No such file or directory [root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]#* Did you build it yourself from Source Code, or did you install someone else's pre-compiled package? If the latter, the packager may have omitted a dependency. It happens from time to time. You probably need to install a - dev or -devel package (what distro are you running?) What do you get for # ldd /usr/sbin/asterisk ? -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-12 issue after successful installation
Hi Team, I have installed asterisk-12 Beta but when I try to asterisk start then get below issue. *[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r asterisk: error while loading shared libraries: libjansson.so.4: cannot open shared object file: No such file or directory [root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]#* -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 12 issue
Hi Team, I have installed asterisk-12 Beta but when I try to asterisk start then get below issue. *[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r asterisk: error while loading shared libraries: libjansson.so.4: cannot open shared object file: No such file or directory [root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]#* -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which events is generated as Asterisk Manager logoff
Hi Team, I am working on Asterisk Events and I am using asterisk 11.4 right now. I want to know which events is regenerates(activate) when asterisk manager logoff from asterisk. I saw *fullybooted *was active when I login into asterisk as manager but no event at logoff time i found. -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 Software Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which events is generated as Asterisk Manager logoff
Hi No even logoff i received in asterisk 11.Might be this is bug or else not sure On Mon, Oct 14, 2013 at 3:10 PM, jg webaccou...@jgoettgens.de wrote: When you logoff yourself, then you send the Logoff Action. Asterisk answers with the response Goodbye and, being a polite dolphin, it thanks for all the fish. If Asterisk shuts down, it sends a Shutdown event. You would still have to monitor the state of the socket connection, in case something else happens. jg -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which events is generated as Asterisk Manager logoff
Okay But I looking for an events which will invoke by asterisk when asterisk manager lost connection. My problem is: I am working on events programming and it's work perfect but sometime don't know why asterisk manager connection lost, So i am looking for an events of logoff of asterisk manager so that I will reconnect asterisk manager again back,,, On Mon, Oct 14, 2013 at 6:12 PM, jg webaccou...@jgoettgens.de wrote: There is no such thing as a Logoff event. You issue a Logoff action and get a reponse. Only if Asterisk is shutting down, there will be a Shutdown event. It looks like this ( denotes what you a sending, and what you will receive from Asterisk): Action: Logoff ActionID: 4711 Response: Goodbye ActionID: 4711 Message: Thanks for all the fish. jg -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which events is generated as Asterisk Manager logoff
Thanks for reply but my question is still not resolved. I will use other method. My connection work 4-6days easily but I am looking for route case of problem On Mon, Oct 14, 2013 at 9:21 PM, jg webaccou...@jgoettgens.de wrote: As I said: - Do it yourself (Action Logoff) - Process the Shutdown event - control the state of the socket I have Manager sessions running for hours without any connection losses. So if you have connection losses then there is likely something else. jg -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which events is generated as Asterisk Manager logoff
As I said, I am running a event capture program and it looks for Events and work on the basis of events. But some time it stop working so I want to auto-connect with asterisk back as it was disconnect with asterisk AMI. On Mon, Oct 14, 2013 at 9:46 PM, jg webaccou...@jgoettgens.de wrote: Can you describe your problem? -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Facing issue in installation of asterisk ...
Virendra Bhati +91-9250078532 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Email to Fax solution
please check it. might be it will help http://ictfax.org/content/installation-guide On Tue, Aug 14, 2012 at 7:20 PM, Ahmed Munir ahmedmunir...@gmail.comwrote: Hi, I would like to know, anyone who worked in Email to Fax scenario? If so please share the idea for implementing it. As on other hand I configured Asterisk for inbound Fax which is working good i.e. later forward the fax via email but don't know how can I implement for outbound fax in this case. Please advice. -- Regards, Ahmed Munir Chohan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718500594 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Background, Playback wave files in asterisk
please read CHANNEL variable. it will help you in this case... On Tue, Aug 7, 2012 at 4:01 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; I discover that I have to place the wave files in the /var/lib/asterisk/sounds/custom/ So, can I understand that the only solution I have is to copy the files that are existed in the path /var/lib/asterisk/sounds/en/ to the path /var/lib/asterisk/sounds/custom? Or there is any other solution? I am using FreePBX and the asterisk version is: Asterisk 1.8.11-cert1 Any advise? Regards Bilal - Hello; What is the difference between using the Background Playback in Asterisk 1.8 without cert and Asterisk 1.8 cert? I surprised that in cert version, I do not hear the sound ! And it is not working properly, but in the normal version, it is working. So what is the new? Is it the version? Or there are some variables or settings need to be done in asterisk 1.8 cert that was not require in the normal version (not cert)? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718500594 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] best free fax solution with asterisk
Thank you all, for your information about FAX.. I will try to used ICTFax and will update you Mr. Tahir Almas.. On Mon, Aug 13, 2012 at 10:58 AM, tahir almas ta...@ictinnovations.comwrote: I will recommend to give ICTFAX http://www.ictfax.org a chance , ICTFAX is based on spandsp and old version work with asterisk http://www.ictfax.org Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Mon, Aug 13, 2012 at 7:08 AM, Bryant Zimmerman brya...@zktech.comwrote: James is this inbound or outbound faxing that is running at 95%. We see about 94% success on inbound faxes, but were not satisfied with that so we started doing some research into the issue to find that the bulk of the fails were actually voice calls, or robo dialers calling fax numbers. Once we threw out those we get about 98% The last 2% include some calls that might not have been faxes, but we were not able to eliminate all of them. Out bound runs at about 98% success. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Steve Underwood ste...@coppice.org *Sent*: Sunday, August 12, 2012 3:56 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] best free fax solution with asterisk On 08/12/2012 10:32 AM, James Sharp wrote: On 8/11/2012 8:05 AM, virendra bhati wrote: Hi team, I want to configure fax with asterisk. there a lot of fax link i found by google but not working perfectly. my setup as follow asterisk 10.x centos 5.8 Want to used T.38 with SpanDSP... Please suggest me the best way. and how to test FoIP ? I use Asterisk 10.3.1, SpanDSP 0.0.6, and Ubuntu 11.10 connecting to Gafachi.com. It works with probably 95% success rate talking via T.38. 95% is pretty bad. Do you know if the failures are mostly during the initial negotiation, or somewhere in the actual FAX exchange? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718500594 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk realtime database structure
best link for asterisk realtime is below one http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example On Fri, Aug 3, 2012 at 1:51 PM, Leandro Dardini ldard...@gmail.com wrote: If you check the contrib/realtime/mysql directory in the source tree, you'll find scripts for almost all the tables. Leandro 2012/8/3 Daniel-Constantin Mierla mico...@gmail.com Hello, I was wondering if there is a tool that can create the realtime database structure for latest Asterisk version or a web resource/file containing the sql scripts. Hope I haven't missed obvious things, I had no luck searching on the web, in the wiki I found few pages with bits of sql or table structures, like: https://wiki.asterisk.org/**wiki/display/AST/SIP+Realtime,** +MySQL+table+structurehttps://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure https://wiki.asterisk.org/**wiki/display/AST/ODBC+**Voicemail+Storagehttps://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage I have several table structures from the Asterisk 1.6, I dug for them in the code or found on the web when I wrote the tutorial about integration with Kamailio 3.1 (http://kb.asipto.com/**asterisk:realtime:kamailio-3.* *1.x-asterisk-1.6.2-astdbhttp://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb), but hopefully now it is an easy way to get the db structure. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- __**__** _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
It means ... Asterisk don't make any IVR at realtime. It just fire Mysql/Odbc query and get *app and appdata.* On Fri, Aug 3, 2012 at 11:50 AM, Leandro Dardini ldard...@gmail.com wrote: It is reasonable 'n' is not usable as priority number. How can asterisk know the second priority if all other priority have 'n' as priority number? In a relational database there is no 'sequential read'. In other words, you need to assign the priority to all entries. Leandro Il giorno 03/ago/2012 06:27, virendra bhati virbh...@gmail.com ha scritto: Hi Team, I want to used *'n*' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
Hi Team, I want to used *'n*' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime issue after registering withx-lite
strange last night my serve had this issue but when next morning i check with register 1000 sip account no issue has come thanks for your reply On Fri, Jul 27, 2012 at 1:30 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Can you please show the database entry for that peer then? On Thu, 2012-07-26 at 23:20 +0530, virendra bhati wrote: My sip.conf don't have any entry related to sip pees. I have everything into database. for more details please check below url, which have good example of asterisk realtime http://bahjons.com/stuff/asterisk-realtime-installation-guide On Thu, Jul 26, 2012 at 11:14 PM, motty.cruz motty.c...@gmail.com wrote: can you post your sip.conf for Exten. 1000? it does not seem like you have [1000] mailbox=1000@default Thanks, -motty __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Thursday, July 26, 2012 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Realtime issue after registering withx-lite Hi All, I have an small issue, which is not creating any problem on working syatem but not sure about the problem that is why eager to know about it. I had installed Asterisk realtime with Asterisk 1.4.41. Every thing is working good but getting warning at Asterisk CLI. [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 Really destroying SIP dialog '0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method: SUBSCRIBE [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 Really destroying SIP dialog '9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method: SUBSCRIBE If anyone have any suggestion please reply to me. -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth
Re: [asterisk-users] What TTS to use?
There are lot of TTS it's depends on you which one you like, flite festival google swift main things of TTS is it's Voice accent. On Thu, Jul 26, 2012 at 3:38 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm thinking about deploying TTS onto our asterisk servers and was just wondering which ones people use and like... Thanks Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime issue after registering with x-lite
Hi All, I have an small issue, which is not creating any problem on working syatem but not sure about the problem that is why eager to know about it. I had installed Asterisk realtime with Asterisk 1.4.41. Every thing is working good but getting warning at Asterisk CLI. [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 Really destroying SIP dialog '0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method: SUBSCRIBE [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 Really destroying SIP dialog '9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method: SUBSCRIBE If anyone have any suggestion please reply to me. -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime issue after registering withx-lite
My sip.conf don't have any entry related to sip pees. I have everything into database. for more details please check below url, which have good example of asterisk realtime http://bahjons.com/stuff/asterisk-realtime-installation-guide On Thu, Jul 26, 2012 at 11:14 PM, motty.cruz motty.c...@gmail.com wrote: ** can you post your sip.conf for Exten. 1000? it does not seem like you have [1000] mailbox=1000@default ** *Thanks, * *-motty* -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati *Sent:* Thursday, July 26, 2012 10:35 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Asterisk Realtime issue after registering withx-lite Hi All, I have an small issue, which is not creating any problem on working syatem but not sure about the problem that is why eager to know about it. I had installed Asterisk realtime with Asterisk 1.4.41. Every thing is working good but getting warning at Asterisk CLI. [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 Really destroying SIP dialog '0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method: SUBSCRIBE [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 Really destroying SIP dialog '9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method: SUBSCRIBE If anyone have any suggestion please reply to me. -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] touch command not behaving for future calls in asterisk 1.4.41
Hi All, It's small issue but making a big problem for my application. I have CentOS release 5.8 (Final) with asterisk 1.4.41 installed. I am using 1.4.41 because Flite work in this version. problem is that when I make changes on .call file to make it future call file with *touch *command then it not changed. [root@server tmp]# touch -t 201207052137 1341509545.39.call [root@server tmp]# ll -rw-r--r-- 1 root root 52 Jul 5 2012 1341509545.39.call .call file's time is missed with year only that's asterisk make call after move to outgoing folder. please give your suggestion. If I am wrong then correct me ... -- Thanks and regards Virendra Bhati +91-9718300881 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] touch command not behaving for future calls in asterisk 1.4.41
Thanks Gohar, I found the issue was copy file to outbound folder not moving. that's why after making future time asterisk start reading file. On Fri, Jul 6, 2012 at 11:16 AM, SamyGo govoi...@gmail.com wrote: Hi, Did you get anything working on it !! See the permission for the user running asterisk process and see if that user can touch files like that. Regards, Sammy On Thu, Jul 5, 2012 at 10:47 PM, virendra bhati virbh...@gmail.comwrote: Hi All, It's small issue but making a big problem for my application. I have CentOS release 5.8 (Final) with asterisk 1.4.41 installed. I am using 1.4.41 because Flite work in this version. problem is that when I make changes on .call file to make it future call file with *touch *command then it not changed. [root@server tmp]# touch -t 201207052137 1341509545.39.call [root@server tmp]# ll -rw-r--r-- 1 root root 52 Jul 5 2012 1341509545.39.call .call file's time is missed with year only that's asterisk make call after move to outgoing folder. please give your suggestion. If I am wrong then correct me ... -- Thanks and regards Virendra Bhati +91-9718300881 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] add new sip account in sip.conf with API Action UpdateConfig with php
Hi List,
I am trying to add new SIP account in new file additional_sip.conf. I read
in Wiki there is API command UpdateConfig which is used to update , add and
delete any entry from configure files. I am using PHP to make new entry in
additional_sip.conf. Below is the code which I tryed
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: admin\r\n);
fputs($socket, Secret: admin\r\n\r\n);
fputs($socket, Action: UpdateConfig\r\n);
fputs($socket, reload=yes\r\n);
fputs($socket, SrcFilename: additional_sip.conf\r\n);
fputs($socket, DstFilename: additional_sip.conf\r\n);
fputs($socket, Action-00: NewCat\r\n);
fputs($socket, Cat-00: 9911881985\r\n);
fputs($socket, Var-00: 9911881985\r\n);
fputs($socket, Value-00: 9911881985\r\n);
fputs($socket, ActionID: 343434\r\n\r\n);
fputs($socket, Action: Logoff\r\n);
fputs($socket, UserName: root\r\n);
fputs($socket, Secret: energy\r\n\r\n);
$done=1;
}
?
*CLI Log:-*
ks3098819*CLI
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
ks3098819*CLI
--
Thanks and regards
Virendra Bhati
+91-08885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
Hyderabad(India)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] add new sip account in sip.conf with API Action UpdateConfig with php
I have update sammy but no luck
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: admin\r\n);
fputs($socket, Secret: admin\r\n\r\n);
fputs($socket, Action: UpdateConfig\r\n);
fputs($socket, reload=yes\r\n);
fputs($socket, SrcFilename: /etc/asterisk/additional_sip.conf\r\n);
fputs($socket, DstFilename: /etc/asterisk/additional_sip.conf\r\n);
fputs($socket, Action-00: NewCat\r\n);
fputs($socket, Cat-00: 9911881985\r\n);
fputs($socket, Var-00: 9911881985\r\n);
fputs($socket, Value-00: 9911881985\r\n);
fputs($socket, ActionID: 343434\r\n\r\n);
fputs($socket, Action: Logoff\r\n);
fputs($socket, UserName: admin\r\n);
fputs($socket, Secret: admin\r\n\r\n);
$done=1;
}
?
On Mon, May 21, 2012 at 5:49 PM, SamyGo govoi...@gmail.com wrote:
Hi,
1- try putting absolute filepath in source and destination field.
2- verify that the permissions of the files you're changing.
Regards,
Sammy.
On Mon, May 21, 2012 at 5:10 PM, virendra bhati virbh...@gmail.comwrote:
Hi List,
I am trying to add new SIP account in new file additional_sip.conf. I
read in Wiki there is API command UpdateConfig which is used to update ,
add and delete any entry from configure files. I am using PHP to make new
entry in additional_sip.conf. Below is the code which I tryed
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: admin\r\n);
fputs($socket, Secret: admin\r\n\r\n);
fputs($socket, Action: UpdateConfig\r\n);
fputs($socket, reload=yes\r\n);
fputs($socket, SrcFilename: additional_sip.conf\r\n);
fputs($socket, DstFilename: additional_sip.conf\r\n);
fputs($socket, Action-00: NewCat\r\n);
fputs($socket, Cat-00: 9911881985\r\n);
fputs($socket, Var-00: 9911881985\r\n);
fputs($socket, Value-00: 9911881985\r\n);
fputs($socket, ActionID: 343434\r\n\r\n);
fputs($socket, Action: Logoff\r\n);
fputs($socket, UserName: root\r\n);
fputs($socket, Secret: energy\r\n\r\n);
$done=1;
}
?
*CLI Log:-*
ks3098819*CLI
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
ks3098819*CLI
--
Thanks and regards
Virendra Bhati
+91-08885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
Hyderabad(India)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-08885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
Hyderabad(India)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Dahdi, libpri of different versions in one pc
when you installed DAHDI/Zaptel on VM then it will work On Mon, Mar 19, 2012 at 4:35 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: I am not sure whether my PRI / BRI card would detect in virtual machine. I have to check. On Sun, Mar 18, 2012 at 8:14 AM, virendra bhati virbh...@gmail.comwrote: you may installed different version at different virtual machines... it will be easy and not time consuming as well. On Wed, Mar 14, 2012 at 11:22 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Its because the card what I have only work with 1.4 and 1.6. On Wed, Mar 14, 2012 at 4:05 AM, John Novack jnov...@stromberg-carlson.org wrote: ** Why would you want to even bother testing EOL products, such as 1.4x and 1.6.x.x? Although I am a 1.4 Luddite, I really don't quite understand why you can't test with 1.8.x or 10, where you mihgt have a hope of getting something fixed if there is a problem, unless you already KNOW there is an issue with later versions. JMO John Novack Gopalakrishnan N wrote: Hi, I would like to install Dahdi, libpri and Asterisk of different versions in one machine. Lets say, Asterisk 1.6.X, Asterisk 1.8.x and Asterisk 1.4.x to be installed in one machine, this can be done using prefix while building configure. For dahdi, libpri can it be done in same way? Because I need to test telephony cards (PRI, BRI, GSM Transcoding) with different versions of Asterisk, libpri and Dahdi, I can't remove and install again of each versions since it is time consuming, sicne there are lot of versions available. Any comments would be appreciated. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 Hyderabad(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 Hyderabad(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Dahdi, libpri of different versions in one pc
you may installed different version at different virtual machines... it will be easy and not time consuming as well. On Wed, Mar 14, 2012 at 11:22 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Its because the card what I have only work with 1.4 and 1.6. On Wed, Mar 14, 2012 at 4:05 AM, John Novack jnov...@stromberg-carlson.org wrote: ** Why would you want to even bother testing EOL products, such as 1.4x and 1.6.x.x? Although I am a 1.4 Luddite, I really don't quite understand why you can't test with 1.8.x or 10, where you mihgt have a hope of getting something fixed if there is a problem, unless you already KNOW there is an issue with later versions. JMO John Novack Gopalakrishnan N wrote: Hi, I would like to install Dahdi, libpri and Asterisk of different versions in one machine. Lets say, Asterisk 1.6.X, Asterisk 1.8.x and Asterisk 1.4.x to be installed in one machine, this can be done using prefix while building configure. For dahdi, libpri can it be done in same way? Because I need to test telephony cards (PRI, BRI, GSM Transcoding) with different versions of Asterisk, libpri and Dahdi, I can't remove and install again of each versions since it is time consuming, sicne there are lot of versions available. Any comments would be appreciated. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 Hyderabad(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where can I find some good examples of listening to AMI events via PHP how to listen to a specific event?
hi, it will help you .. http://www.micpc.com/eventmonitor/ On Fri, Feb 24, 2012 at 9:38 AM, Ast Coder asteriskcod...@gmail.com wrote: Hi everyone, I got HTTP AMI working fine here. For example this dials 1-415-999-and then sends to Extension @from-internal : http://192.168.0.100:8088/asterisk/manager?action=commandoriginateDAHDI/g0/1415999extension@from-internal However, I want to have some control over this call. I want to be notified the moment this call is hangup. I guess there would be a hangup event generated. I am not sure if that would be done through action:waitevent? or if there is another method. I am also looking for some php samples on listening for these events as I am trying to create a Web GUI for a dialer that will allow me to show status of a call in real-time like Call In Progress, Call Ended, etc... I see that too many events are generated and I am wondering if there is an easy way of listening for a particular event? Would that be ActionID? if so, how to use it? Thanks a lot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
Does anyone know the correct information of my question. All are move round and round . On Tue, Feb 21, 2012 at 7:28 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/21/2012 07:51 AM, Alex Balashov wrote: As many ports as required by the nature of the call, i.e. the protocol(s) used for the bearer. For an IAX2 call, the answer is 'zero' for all of those call types (at least the ones that are supported in IAX2, not all of them are). For protocols that use RTP for media transport, two ports are required for each media stream (one for RTP, one for RTCP). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
Hi Kevin, I appreciate, that you replyed first and even fast. But you asked for nature of call too. So my question was added with your question . Voice call *How many port of UDP or RTP ?* Video call *How many port of UDP or RTP ?* Fax call*How many port of UDP or RTP ?* T.140 text call* How many port of UDP or RTP ?* As per the reply Voice call is come to 4 ports but rest is not clear. Voice call *4* Video call *??* Fax call*??* T.140 text call* ??* *Will these port of UDP, RPT or Both ?* On Wed, Feb 22, 2012 at 6:08 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/22/2012 06:26 AM, virendra bhati wrote: Does anyone know the correct information of my question. All are move round and round . What does that mean? I answered your question with the correct and complete information. On Tue, Feb 21, 2012 at 7:28 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 02/21/2012 07:51 AM, Alex Balashov wrote: As many ports as required by the nature of the call, i.e. the protocol(s) used for the bearer. For an IAX2 call, the answer is 'zero' for all of those call types (at least the ones that are supported in IAX2, not all of them are). For protocols that use RTP for media transport, two ports are required for each media stream (one for RTP, one for RTCP). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
thanks for suggesting the link. Yes i don't have networking, and good SIP communication knowledge. On Wed, Feb 22, 2012 at 6:41 PM, Phil Frost p...@macprofessionals.comwrote: On 02/22/2012 08:01 AM, virendra bhati wrote: *Will these port of UDP, RPT [assume you mean RTP] or Both ?* It's evident from your response that you do not have a solid understanding of networking fundamentals. The full answer to your question will quickly go out of scope of this list and become an introduction to IP fundamentals. So, I suggest you start by reading these: http://en.wikipedia.org/wiki/**OSI_modelhttp://en.wikipedia.org/wiki/OSI_model http://en.wikipedia.org/wiki/**Internet_Protocolhttp://en.wikipedia.org/wiki/Internet_Protocol http://en.wikipedia.org/wiki/**User_Datagram_Protocolhttp://en.wikipedia.org/wiki/User_Datagram_Protocol http://en.wikipedia.org/wiki/**Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Real-time_Transport_Protocol -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how many UDP ports is required for 1 call
Hi, how many UDP ports is required for 1 call. and why . -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
right now it's only voice call. But thanks for segregate the call. Now i want to know about all calls used port too. On Tue, Feb 21, 2012 at 7:06 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/21/2012 07:30 AM, virendra bhati wrote: Hi, how many UDP ports is required for 1 call. and why . A 'call' is too ambiguous to answer your question. Is this a voice call, a video/voice call, a FAx call, a T.140 text call, or something else? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] India Pune Pri call problem
Satish, As if I know, PRI provider give you PRI number at the time of purchase and even billing documents will be made on the basis of the number only. So how you can set another Caller-id number for that allotted number. But you can do only change the PRI number for outside world after discussion with PRI provider. I did the same with Idea UP West circle. They provide me 3 Callerid for single PRI lines for making OBD calls on that circle. So all things is depends on PRI providers not at your end. On Tue, Feb 14, 2012 at 1:24 PM, Satish Barot satish4aster...@gmail.comwrote: Indian Telcos do allow setting callerid on PRI line and you can set the callerid to one of the numbers allocated by them for PRI. --Satish Barot On Mon, Feb 13, 2012 at 6:49 PM, Ast Coder asteriskcod...@gmail.comwrote: India TRAI rules doesn't allow for CLID setting. They are backwards minded. If you ever get them to do it let me know ;) -Bruce On Mon, Feb 13, 2012 at 8:18 AM, Steven Howes steve-li...@geekinter.netwrote: On 13 Feb 2012, at 12:06, virendra bhati wrote: You can't set callerid for outgoing calls in case of PRI. Why not? Every PRI I have used supported it. Is this a carrier-specific thing? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
Thanks for reply and share your techniques, dialplans and knowledge on this thread. But my question was not related to load-balancing. I want to know , Why freeSwitch can preferred with compare to Asterisk(Call base , quality base)? And what is architecture difference between them. I am totally agree that by using SIPp we can not relay that server can handle so much load. because by using MOH only CPU load can major and we can check how many thread asterisk can open. On Fri, Feb 10, 2012 at 2:34 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/09/2012 01:17 PM, Danny Nicholas wrote: If the MOH thing is really true, a more realistic test would be to run playback(demo-instruct). Since I know that I will eventually cross this bridge in real life/real time, I devised this test on my Asterisk 10.0 box Dialplan (in default context) exten = 3366,1,answer() exten = 3366,n,playback(demo-instruct,**noanswer) exten = 3366,n,playback(demo-instruct,**noanswer) exten = 3366,n,playback(vm-goodbye,**noanswer) exten = 3366,n,hangup() SIPP command ./sipp -l 399 -d 99000 -m 399 -s 3366 -p 5061 -sn uac 127.0.0.1 -trace_err I was able to do 260 concurrent calls with no issues. The 2 playbacks for demo-instruct were to cover 99 seconds since the file is only 67 seconds long. For the 300/1000 call scenario, you would need to duplicate the line accordingly. The limiting factor for me was my rtp.conf. I set up a range of 10001-10520 which stopped at 260 since each call allocates 4 rtp slots (2 in use and 2 for transfer, etc). That's not quite correct. RTP ports are not allocated for 'transfers'. 2 ports are used for each media stream that can be used on a channel. Since each channel has an audio stream, that will consume 2 ports. If video support is enabled for the channel (even if it is not in use), then 2 more ports will be consumed. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk V/s FreeSwitch
Hi List, Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
yes concurrent calls(CC). On Tue, Feb 7, 2012 at 5:27 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: You mean concurrent calls? You can have several 100 concurrent calls with a good CPU in newer versions of asterisk, however calls per secons (CPS) have some limitations I guess reason being that both are different in Architecture, Asterisk was designed keeping PBX in mind but Freeswitch was for SIP switching Regards, Zohair Raza On Tue, Feb 7, 2012 at 3:38 PM, virendra bhati virbh...@gmail.com wrote: Hi List, Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
thanks Gilles, After reading these web links. it's pretty clear that FreeSwitch is batter then Asterisk feature, quality wise. But asterisk is easy to used. But the question is still open from my end. *How* *FreeSwitch can support 1000CC but asterisk not* ? Because FreeSwitch used XML as configuration and asterisk plan text file ? FreeSwitch used sofia_sip and asterisk used sip ? Asterisk is PBX and FreeSwitch is SoftSwitch ? On Tue, Feb 7, 2012 at 9:10 PM, Gilles codecompl...@free.fr wrote: On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.com wrote: Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... Provided Asterisk, even in release 1.8 or 10, does handle much fewer concurrent calls than Freeswitch, you might find the answer in those articles: How does FreeSWITCH compare to Asterisk? www.freeswitch.org/node/117 Asterisk vs FreeSWITCH www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ Asterisk vs. FreeSWITCH www.anders.com/cms/266 Open Source VoIP: Asterisk or FreeSwitch? www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233 FreeSwitch vs Asterisk www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can someone tell me what is this issue ?
Call is not routing from server to destination. app8*CLI console dial 00918885268942 [Feb 3 06:01:15] NOTICE[28124]: console_video.c:133 console_video_start: voice only, console video support not present -- Executing [00918885268942@default:1] Answer(Console/dsp, ) in new stack Console call has been answered -- Executing [00918885268942@default:2] Dial(Console/dsp, SIP/00918885268942@voipon) in new stack == Using SIP RTP CoS mark 5 Audio is at 10.30.131.136 port 12556 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 217.14.138.127:5065: INVITE sip:00918885268...@sip.voipon.co.uk:5065;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.30.131.136:5060;branch=z9hG4bK5388007b;rport Max-Forwards: 70 From: asterisk sip:7476...@sip.voipon.co.uk;tag=as2f61c90c To: sip:00918885268...@sip.voipon.co.uk:5065;user=phone Contact: sip:7476849@10.30.131.136 Call-ID: 3cd12da658b42c10186c01ed3a7d2...@sip.voipon.co.uk CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.21 Date: Fri, 03 Feb 2012 06:01:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 313 v=0 o=root 1850926672 1850926672 IN IP4 10.30.131.136 s=Asterisk PBX 1.6.2.21 c=IN IP4 10.30.131.136 t=0 0 m=audio 12556 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 00918885268942@voipon Retransmitting #1 (NAT) to 217.14.138.154:5060: INVITE sip:00918885268...@sip.voipon.co.uk:5065;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.30.131.136:5060;branch=z9hG4bK5388007b;rport Max-Forwards: 70 From: asterisk sip:7476...@sip.voipon.co.uk;tag=as2f61c90c To: sip:00918885268...@sip.voipon.co.uk:5065;user=phone Contact: sip:7476849@10.30.131.136 Call-ID: 3cd12da658b42c10186c01ed3a7d2...@sip.voipon.co.uk CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.21 Date: Fri, 03 Feb 2012 06:01:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 313 Scheduling destruction of SIP dialog ' 3cd12da658b42c10186c01ed3a7d2...@sip.voipon.co.uk' in 32000 ms (Method: INVITE) -- SIP/voipon-0014 is circuit-busy Scheduling destruction of SIP dialog ' 3cd12da658b42c10186c01ed3a7d2...@sip.voipon.co.uk' in 32000 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/1/0) -- Executing [00918885268942@default:3] NoOp(Console/dsp, **CONGESTION**) in new stack -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] read digits during recording / DTMF in conference?
You may used even capturing in the case... when call is recoding in conference On Wed, Feb 1, 2012 at 4:04 PM, Kingsley Tart kings...@skymarket.co.ukwrote: Hi, I want to create a system for incoming calls where, under some circumstances, callers get routed straight to voicemail (or some other means of recording a message) but if they enter a valid extension number then the recorded message would be abandoned and they'd be diverted to the extension number they entered. I realise this can be done with the voicemail app with operator=yes but the problem with this is that the caller has to press 0 while the announcement is being played. If they're too slow and recording has started, they've missed the opportunity. So I played around with ConfBridge and a couple of call files, just to see if I could get it to work. It's a bit convoluted but the idea is that the caller gets silently put into a conference, then two call files make asterisk silently connect to other calls into the same conference, with one doing the recording and the other using Read() to collect digits. If I just had the caller and one of the other calls in the conference (the one doing Read()) then this worked - Read() managed to read the DTMF digits and assign them to a variable. However, when the 'recording' call is also in the conference, the 'read' call can no longer recognise the DTMF digits. To test, I made the 'read' call play a sound before calling Read() and I could hear this being played so the call was definitely there. However, regardless of the number of digits I pressed, Read() didn't notice any of them, even if I introduced a delay so that the other channels were quiet before the call to Read(). I realise this might seem a bit like a mad solution but can anyone else think of a way to get Asterisk to read (and react to) DTMF digits during a recording? This is with Asterisk 1.8.7. -- Cheers, Kingsley. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange how Asterisk know the updated information of log
Logger rotate is used to reload and start asterisk log of Events and quesue. And I want to store the complete log of asterisk for day-to-day report. that's y want to store in another file for future purpose. On Fri, Jan 27, 2012 at 2:12 PM, Alec Davis siva...@paradise.net.nz wrote: I want to make a new file of CLI log everyday. So I just make a shell script in asterisk log directory. My file is working fine and making new file with the name of full_2012-01-27. But strange I noticed that asterisk is updating my newly crested files even i don't reload asterisk. The CLI command 'logger rotate' may be a better way. Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange how Asterisk know the updated information of log
Hi, Doing some changes on logger.conf and with the help of cli logger rotate now problem is solved. thank you Alec.. On Fri, Jan 27, 2012 at 2:35 PM, virendra bhati virbh...@gmail.com wrote: Logger rotate is used to reload and start asterisk log of Events and quesue. And I want to store the complete log of asterisk for day-to-day report. that's y want to store in another file for future purpose. On Fri, Jan 27, 2012 at 2:12 PM, Alec Davis siva...@paradise.net.nzwrote: I want to make a new file of CLI log everyday. So I just make a shell script in asterisk log directory. My file is working fine and making new file with the name of full_2012-01-27. But strange I noticed that asterisk is updating my newly crested files even i don't reload asterisk. The CLI command 'logger rotate' may be a better way. Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange how Asterisk know the updated information of log
Hi All, I want to make a new file of CLI log everyday. So I just make a shell script in asterisk log directory. My file is working fine and making new file with the name of *full_2012-01-27*. But strange I noticed that asterisk is updating my newly crested files even i don't reload asterisk. So how asterisk know that file name is changed ? why not asterisk make new file with the name of *full* ? Can someone please tell me this behaviour of Asterisk (1.6.2.20). -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] View # active calls in a context
make group with the name of context and you may get the current calls on the basis of these groups On Sat, Jan 21, 2012 at 6:51 PM, Michelle Dupuis mdup...@ocg.ca wrote: We have a multitenant Asterisk 1.4 installation for multiple small business, and we need to report how many calls a single business has active at one time. Is there a way to VIEW how many calls are up in a single context? (Or some other way to accomplish the same)? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force CDR to be written.
yes , you can do it but you need your own script which will fill the databases tables. On Sat, Jan 21, 2012 at 9:21 PM, Jim DeVito asterisk-users-mailing-l...@devito.cc wrote: Is there a way to Force the CDR data to be written prior to Hanging up the channel? Thanks!! Jim -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to support Dialogic Cards
Kevin, Dialogic doesn't provide any soultion as open source. It provides hardware base cards for making outbond calls. And they used asterisk as backend for they card application. On Thu, Jan 19, 2012 at 6:50 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 01/18/2012 11:14 PM, virendra bhati wrote: Yes you may used Dialogic card with asterisk. but it's depends on the requirements too. I'm not sure what that response is supposed to mean... I can't really parse it. If you want to use Dialogic cards with Asterisk, you'll need to contact Dialogic about getting an Asterisk channel driver module for them. To my knowledge there is no open-source channel driver available for any Dialogic cards. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failed to Allocate port for RTP instance
Hi have you open the port in rtp.conf ? rtpstart=1 rtpend=2 On Wed, Jan 18, 2012 at 1:14 PM, shalu dhamija shalu.dham...@rancoretech.com wrote: Hello, I am trying to deposit a voicemail message(using voicemail() application) for a subscriber using asterisk-1.8.7.1. But i am facing aproblem in the rtp port allocation for a session due to which '488 Not Acceptable' response is sent towards the client end. Following are error messages: [Jan 18 12:43:59] ERROR[19164] res_rtp_asterisk.c: Failed to Allocate port 7660 for RTP instance '0x1a75ab98' [Jan 18 12:43:59] ERROR[19164] res_rtp_asterisk.c: Oh dear... we couldn't allocate a port (x=7662)7660 for RTP instance '0x1a75ab98'. errno 99 [Jan 18 12:43:59] DEBUG[19164] rtp_engine.c: Engine 'asterisk' failed to setup RTP instance '0x1a75ab98' [Jan 18 12:43:59] DEBUG[19164] rtp_engine.c: Destroyed RTP instance '0x1a75ab98' [Jan 18 12:43:59] DEBUG[19164] chan_sip.c: ERROR: failed to allocate rtp instance [Jan 18 12:43:59] DEBUG[19164] chan_sip.c: Could not initialize RTP instance for dialog: 800E51A5-1140-E111-A216-001A4B4698C3@10.34.77.90 Please find attached the log file for more information. Regards, Shalu -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to support Dialogic Cards
Yes you may used Dialogic card with asterisk. but it's depends on the requirements too. On Thu, Jan 19, 2012 at 9:05 AM, Vinod Dharashive vdharash...@gmail.comwrote: Hi Team, Is there any way that asterisk can support Dialogic card, i have done lot of search but could find any useful information. Thanks Vinod Dharashive -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prepaid billing
Hi Zohair, By using only asterisk it's not possible. So used progremming languages and do realtime billing at your ends. like 1st caller will take complete amount ($5) and if 2nd call will come then deduct used amount and share remaining amount to others like that. On Tue, Jan 17, 2012 at 9:54 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Hi All, I am writing a billing engine in AGI. My scenario is : One customer can have simultaneous calls and I need to hang up one customer's all call when balance reaches 0 If I set limit for each call using 'L' in dial command, lets say 5 minutes in accordance with remaining credit and connect the call, few seconds later a 2nd call comes in and the first call is still in progress. If I permit the same 5 minutes as per this formula and both calls remains connected for the next 5 minutes then credit will go in minus which is not acceptable. One option is to charge credit via AMI and as soon as the credit goes 0, hangup all calls for this customer. Is there any other way to achieve this ? Regards, Zohair Raza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prepaid billing
Batter is used DB to store intime of call then when ever currect used time is required then deduct from intime - current time. On Wed, Jan 18, 2012 at 1:01 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Hi, I understand this, but I think there isn't any option that helps us to reduce cost while call is in progress. One option that I was thinking is to check elapsed time by core show channel channel-id and deduct the amount but we need to check it every second or x seconds via AMI. Regards, Zohair Raza On Wed, Jan 18, 2012 at 9:35 AM, virendra bhati virbh...@gmail.comwrote: Hi Zohair, By using only asterisk it's not possible. So used progremming languages and do realtime billing at your ends. like 1st caller will take complete amount ($5) and if 2nd call will come then deduct used amount and share remaining amount to others like that. On Tue, Jan 17, 2012 at 9:54 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hi All, I am writing a billing engine in AGI. My scenario is : One customer can have simultaneous calls and I need to hang up one customer's all call when balance reaches 0 If I set limit for each call using 'L' in dial command, lets say 5 minutes in accordance with remaining credit and connect the call, few seconds later a 2nd call comes in and the first call is still in progress. If I permit the same 5 minutes as per this formula and both calls remains connected for the next 5 minutes then credit will go in minus which is not acceptable. One option is to charge credit via AMI and as soon as the credit goes 0, hangup all calls for this customer. Is there any other way to achieve this ? Regards, Zohair Raza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to set callerid in php AGI file.
Hi,
I am using phpagi for agi scripting. and want to update callerid number but
didn't get any success. please help me how to update PHPAGI is new for me.
Below is the code which I write.
#!/usr/bin/php -q
?php
set_time_limit(30);
//require(.phpagi.php.);
include(phpagi.php);
$agi = new AGI();
//answer the call
$agi- answer();
$agi-verbose(--);
$agi- exec('Set',CALLERID(num)=01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi-set_variable(CALLERID(num),01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi- exec('Dial',SIP/00918885268...@sip.trunk.gradwell.com,60,r);
//$agi- exec('Dial',SIP/00918885268942@voipon,60,r);
?
And CLI
== Using SIP RTP CoS mark 5
-- Executing [101@outbound:1] Answer(SIP/2209-26d3, ) in new
stack
-- Executing [101@outbound:2] AGI(SIP/2209-26d3,
/home/virendra.bhati/outdial.php) in new stack
-- Launched AGI Script /home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_request: /home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_channel: SIP/2209-26d3
SIP/2209-26d3AGI Tx agi_language: en
SIP/2209-26d3AGI Tx agi_type: SIP
SIP/2209-26d3AGI Tx agi_uniqueid: 1326357644.10070
SIP/2209-26d3AGI Tx agi_version: 1.6.2.20
SIP/2209-26d3AGI Tx agi_callerid: 2209
SIP/2209-26d3AGI Tx agi_calleridname: unknown
SIP/2209-26d3AGI Tx agi_callingpres: 0
SIP/2209-26d3AGI Tx agi_callingani2: 0
SIP/2209-26d3AGI Tx agi_callington: 0
SIP/2209-26d3AGI Tx agi_callingtns: 0
SIP/2209-26d3AGI Tx agi_dnid: 101
SIP/2209-26d3AGI Tx agi_rdnis: unknown
SIP/2209-26d3AGI Tx agi_context: outbound
SIP/2209-26d3AGI Tx agi_extension: 101
SIP/2209-26d3AGI Tx agi_priority: 2
SIP/2209-26d3AGI Tx agi_enhanced: 0.0
SIP/2209-26d3AGI Tx agi_accountcode:
SIP/2209-26d3AGI Tx agi_threadid: 1386719552
SIP/2209-26d3AGI Tx
SIP/2209-26d3AGI Rx ANSWER
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx VERBOSE
-- 1
/home/virendra.bhati/outdial.php:
--
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx EXEC Set CALLERID(num)=01133200274
-- AGI Script Executing Application: (Set) Options:
(CALLERID(num)=01133200274)
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx SET VARIABLE CALLERID(num) 01133200274
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx EXEC Dial SIP/
00918885268...@sip.trunk.gradwell.com,60,r
-- AGI Script Executing Application: (Dial) Options: (SIP/
00918885268...@sip.trunk.gradwell.com,60,r)
== Using SIP RTP CoS mark 5
ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
-- Called 00918885268...@sip.trunk.gradwell.com
[Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463 handle_response_invite:
Received response: Forbidden from '01133200274
sip:01133200274@10.10.10.181;tag=as76229e88'
-- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
SIP/2209-26d3AGI Tx 200 result=0
-- SIP/2209-26d3AGI Script /home/virendra.bhati/outdial.php
completed, returning 0
-- Executing [101@outbound:3] Hangup(SIP/2209-26d3, ) in new
stack
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] how to set callerid in php AGI file.
How to used it in AGI ? I think it's Dialplan apps.
On Thu, Jan 12, 2012 at 2:18 PM, Zohair Raza
engineerzuhairr...@gmail.comwrote:
Hi,
Try setting CDR(clid)
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati virbh...@gmail.comwrote:
Hi,
I am using phpagi for agi scripting. and want to update callerid number
but didn't get any success. please help me how to update PHPAGI is new for
me. Below is the code which I write.
#!/usr/bin/php -q
?php
set_time_limit(30);
//require(.phpagi.php.);
include(phpagi.php);
$agi = new AGI();
//answer the call
$agi- answer();
$agi-verbose(--);
$agi- exec('Set',CALLERID(num)=01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi-set_variable(CALLERID(num),01133200274);
$ani = $agi-request['agi_callerid'];
$agi-noop(My CalleID: =.$ani);
$agi- exec('Dial',SIP/00918885268...@sip.trunk.gradwell.com,60,r);
//$agi- exec('Dial',SIP/00918885268942@voipon,60,r);
?
And CLI
== Using SIP RTP CoS mark 5
-- Executing [101@outbound:1] Answer(SIP/2209-26d3, ) in new
stack
-- Executing [101@outbound:2] AGI(SIP/2209-26d3,
/home/virendra.bhati/outdial.php) in new stack
-- Launched AGI Script /home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_request: /home/virendra.bhati/outdial.php
SIP/2209-26d3AGI Tx agi_channel: SIP/2209-26d3
SIP/2209-26d3AGI Tx agi_language: en
SIP/2209-26d3AGI Tx agi_type: SIP
SIP/2209-26d3AGI Tx agi_uniqueid: 1326357644.10070
SIP/2209-26d3AGI Tx agi_version: 1.6.2.20
SIP/2209-26d3AGI Tx agi_callerid: 2209
SIP/2209-26d3AGI Tx agi_calleridname: unknown
SIP/2209-26d3AGI Tx agi_callingpres: 0
SIP/2209-26d3AGI Tx agi_callingani2: 0
SIP/2209-26d3AGI Tx agi_callington: 0
SIP/2209-26d3AGI Tx agi_callingtns: 0
SIP/2209-26d3AGI Tx agi_dnid: 101
SIP/2209-26d3AGI Tx agi_rdnis: unknown
SIP/2209-26d3AGI Tx agi_context: outbound
SIP/2209-26d3AGI Tx agi_extension: 101
SIP/2209-26d3AGI Tx agi_priority: 2
SIP/2209-26d3AGI Tx agi_enhanced: 0.0
SIP/2209-26d3AGI Tx agi_accountcode:
SIP/2209-26d3AGI Tx agi_threadid: 1386719552
SIP/2209-26d3AGI Tx
SIP/2209-26d3AGI Rx ANSWER
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx VERBOSE
-- 1
/home/virendra.bhati/outdial.php:
--
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx EXEC Set CALLERID(num)=01133200274
-- AGI Script Executing Application: (Set) Options: (CALLERID(num)=
01133200274)
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx SET VARIABLE CALLERID(num) 01133200274
SIP/2209-26d3AGI Tx 200 result=1
SIP/2209-26d3AGI Rx NOOP My CalleID: =2209
SIP/2209-26d3AGI Tx 200 result=0
SIP/2209-26d3AGI Rx EXEC Dial SIP/
00918885268...@sip.trunk.gradwell.com,60,r
-- AGI Script Executing Application: (Dial) Options: (SIP/
00918885268...@sip.trunk.gradwell.com,60,r)
== Using SIP RTP CoS mark 5
ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
-- Called 00918885268...@sip.trunk.gradwell.com
[Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463
handle_response_invite: Received response: Forbidden from '01133200274
sip:01133200274@10.10.10.181;tag=as76229e88'
-- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
SIP/2209-26d3AGI Tx 200 result=0
-- SIP/2209-26d3AGI Script /home/virendra.bhati/outdial.php
completed, returning 0
-- Executing [101@outbound:3] Hangup(SIP/2209-26d3, ) in new
stack
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http
Re: [asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??
Hi Shalu, How you are invoking call in dialplan. it's completely depends on that. And error show that no voice is there for store in voicemail . On Wed, Jan 11, 2012 at 10:05 AM, shalu dhamija shalu.dham...@rancoretech.com wrote: Hello, I am trying to run load on asterisk server(version 1.8.7.1) for the voicemail() application using SIPp tool. I am just running sipp at call rate of 1 cps with the following command: ./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 --trace_err I am trying to deposit 9000 messages in the mailbox of user 1 (given by the -s option) but the following warning is coming on the asterisk server due to which the message does not get deposited into the users mailbox: No audio available on SIP/172.16.129.13:5060-0001?? I have set rtpstart=6000 and rtpend=2 in rtp.conf. Can someone please let me know how to avoid these kind of warnings. Thanks. Shalu Thanks and Regards, Shalu Dhamija Rancore Technologies(P) Ltd. Gurgaon Ph : 0124-4200691 +91-9910995356(M) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous
Hi, Give the complete details about the asterisk version, and SIP trunk conf details On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous
Hi checked your debug like. Did you check that your SIP device ir registered with server ? if yes then dial below command from CLI *originate sip/test02 application dial* On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hi, I am using asterisk ver 1.8.8.1. My SIP trunk conf details are below.. [general] context=default ; Default context for incoming calls realm=192.168.1.55 allowguest=yes realmauth=yes send_rpid=pai register = test02:test02@192.168.1.55 [test02] type=peer nat=no canreinvite=no host=192.168.1.55 ;realm=test02@192.168.1.55 context=incoming secret=test02 permit=192.168.1.0/255.255.255.0 username=test02 fromuser=test02 fromdomain=192.168.1.55 defaultuser=test02 insecure=invite,port outboundproxy=192.168.1.55 promiscredir=yes userphone=yes For more details you can find my paste in pastebin.. Links given below. While Dialing call fro Xlite send following Sip header F= sip:test02@192.168.1.55. And if tried to register same account in asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I dont know why asterisk sends anonymous.invalid instead of domain name..Help me Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.com wrote: Hi, Give the complete details about the asterisk version, and SIP trunk conf details On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
Hi, Please give you sip phone name and sip.conf and extensions.conf details which is using for that communication. And CLI output of asterisk is also required. On Tue, Jan 3, 2012 at 9:59 AM, Faraj Khasib fkha...@iconnecths.com wrote: I use asterisk 1.6, my clients are sip clients, I dail using audio call in my clients but the request is recieved at the other client as video call request since I am enabling video support for sip Sent from my iPhone On ٠٢/٠١/٢٠١٢, at ١١:٤٩ م, Doug Lytle supp...@drdos.info wrote: Faraj Khasib wrote: Please help, I have tried many things I cannt make it work, when I make an audio call it is converted by asterisk to video call request Not that I can help, since I don't do any video calling. But, if you don't give any information about your system (OS and version, Asterisk version and what type of phone you are using), you're not likely to get much of a response. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
Which is means like if you are using sip 1234 then give the details of [1234] into that open thread and relevent extensions details too On Tue, Jan 3, 2012 at 11:30 AM, Faraj Khasib fkha...@iconnecths.comwrote: Which is?! What I am missing how to set dail plan in extension.conf to pass call type as its Not convert request to video Sent from my iPhone On ٠٣/٠١/٢٠١٢, at ٧:٢٩ ص, virendra bhati virbh...@gmail.com wrote: Hi, Please give you sip phone name and sip.conf and extensions.conf details which is using for that communication. And CLI output of asterisk is also required. On Tue, Jan 3, 2012 at 9:59 AM, Faraj Khasib fkha...@iconnecths.comwrote: I use asterisk 1.6, my clients are sip clients, I dail using audio call in my clients but the request is recieved at the other client as video call request since I am enabling video support for sip Sent from my iPhone On ٠٢/٠١/٢٠١٢, at ١١:٤٩ م, Doug Lytle supp...@drdos.info wrote: Faraj Khasib wrote: Please help, I have tried many things I cannt make it work, when I make an audio call it is converted by asterisk to video call request Not that I can help, since I don't do any video calling. But, if you don't give any information about your system (OS and version, Asterisk version and what type of phone you are using), you're not likely to get much of a response. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
Hi Might be it will help. Read it and set in extension as per your need. core show function CHANNEL -= Info about function 'CHANNEL' =- [Synopsis] Gets/sets various pieces of information about the channel. [Description] Gets/sets various pieces of information about the channel, additional item may be available from the channel driver; see its documentation for details. Any item requested that is not available on the current channel will return an empty string. [Syntax] CHANNEL(item) [Arguments] item Standard items (provided by all channel technologies) are: audioreadformat - R/O format currently being read. * audionativeformat - R/O format used natively for audio.* audiowriteformat - R/O format currently being written. callgroup - R/W call groups for call pickup. channeltype - R/O technology used for channel. language - R/W language for sounds played. musicclass - R/W class (from musiconhold.conf) for hold music. parkinglot - R/W parkinglot for parking. rxgain - R/W set rxgain level on channel drivers that support it. state - R/O state for channel tonezone - R/W zone for indications played transfercapability - R/W ISDN Transfer Capability, one of: SPEECH DIGITAL RESTRICTED_DIGITAL 3K1AUDIO DIGITAL_W_TONES VIDEO txgain - R/W set txgain level on channel drivers that support it. * videonativeformat - R/O format used natively for video* trace - R/W whether or not context tracing is enabled, only available *if CHANNEL_TRACE is defined*. *chan_sip* provides the following additional options: peerip - R/O Get the IP address of the peer. recvip - R/O Get the source IP address of the peer. from - R/O Get the URI from the From: header. uri - R/O Get the URI from the Contact: header. useragent - R/O Get the useragent. peername - R/O Get the name of the peer. t38passthrough - R/O '1' if T38 is offered or enabled in this channel, otherwise '0' rtpqos - R/O Get QOS information about the RTP stream This option takes two additional arguments: Argument 1: 'audio' Get data about the audio stream 'video' Get data about the video stream 'text' Get data about the text stream Argument 2: 'local_ssrc'Local SSRC (stream ID) 'local_lostpackets' Local lost packets 'local_jitter' Local calculated jitter 'local_maxjitter' Local calculated jitter (maximum) 'local_minjitter' Local calculated jitter (minimum) 'local_normdevjitter'Local calculated jitter (normal deviation) 'local_stdevjitter' Local calculated jitter (standard deviation) 'local_count' Number of received packets 'remote_ssrc' Remote SSRC (stream ID) 'remote_lostpackets'Remote lost packets 'remote_jitter' Remote reported jitter 'remote_maxjitter' Remote calculated jitter (maximum) 'remote_minjitter' Remote calculated jitter (minimum) 'remote_normdevjitter'Remote calculated jitter (normal deviation) 'remote_stdevjitter'Remote calculated jitter (standard deviation) 'remote_count' Number of transmitted packets 'rtt' Round trip time 'maxrtt'Round trip time (maximum) 'minrtt'Round trip time (minimum) 'normdevrtt'Round trip time (normal deviation) 'stdevrtt' Round trip time (standard deviation) 'all' All statistics (in a form suited to logging, but not for parsing) rtpdest - R/O Get remote RTP destination information. This option takes one additional argument: Argument 1: 'audio' Get audio destination 'video' Get video destination 'text' Get text destination *chan_iax2* provides the following additional options: peerip - R/O Get the peer's ip address. peername - R/O Get the peer's username. [See Also] Not available On Tue, Jan 3, 2012 at 11:53 AM, Faraj Khasib fkha...@iconnecths.comwrote: Here is the thing, my sip client can call the same. Extension once as audio and once as video, so I cannt turn off video supportat reciever, what I guess can be done is in extension.conf , there must be flag or something I can manipulate ... Sent from my iPhone On ٠٣/٠١/٢٠١٢, at ٨:١٩ ص, virendra bhati virbh...@gmail.com wrote: Which is means like if you are using sip 1234 then give the details of [1234] into that open thread and relevent extensions details too On Tue, Jan 3, 2012 at 11:30 AM, Faraj Khasib fkha...@iconnecths.comwrote: Which is?! What I am missing how to set dail plan in extension.conf to pass call type as its Not convert request to video Sent from my iPhone On ٠٣
Re: [asterisk-users] performance/memory
Hi, http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sorting Read it, Might be it wil solved your doubts. On Fri, Dec 30, 2011 at 12:01 AM, Matt Hamilton mistral9...@hotmail.comwrote: I have a couple of performance/memory related questions: Is there any downside to using long URIs as far as memory or database (mysql) performance is concerned, e.g. sip:1234567890_1234567...@abc.com? Or is this negligible? Also is there a performance hit if no pattern matching is used? e.g. exten = _XXX,Noop(... vs exten = 100,Noop(.. exten = 101,Noop(... exten = 102,Noop(... ... exten = 999,Noop(... If a call comes to 999, does Asterisk go through each extension sequentially from 100 to 999 until it finds the matching one? Thanks, Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Testing software to test IVR system
In server B if I use SendDTMF then it means I am changing programming at
server B. Actually I don't have right or permission to change programming
in server B.
otherwise your suggestion is best for channel base communication.
On Thu, Dec 29, 2011 at 2:33 PM, Sammy Govind govoi...@gmail.com wrote:
Easy, use Read() to capture the incoming DTMF from Server-B
Server-A Server-B
Initiate-Call - AnswerCall()
SendDTMF(5)-- Read()
Read()-SendDTMF(4)
SendDTMF(3)-- Read()
Read()-SendDTMF(2)
SendDTMF(1)-- Read()
Put proper GOTOIFs after reads if you like.
--
Regards,
Sammy
On Thu, Dec 29, 2011 at 12:34 PM, virendra bhati virbh...@gmail.comwrote:
I originate calls from .call file and 1 channel I have at A server A and
another channel at B server.
*A server code is below:-*
exten = 43689956,1,Answer()
same = n,Wait(5)
same = n,SendDTMF(1)
same = n,NoOp(== ${CHANNEL(state)}== state)
same = n,wait(2)
same = n,SendDTMF(123456789012345#)
same = n,NoOp(== ${CHANNEL(state)}== state)
same = n,Hangup()
_ _
| A server | ___DTMF Send_= | B server |
|_| =--- Responce - |_|
*B server code is below:-*
At B server call come to 201 extension which is mention here..
exten = _20[1-6],1,Answer()
same = n,Ringing()
same = n,wait(2)
same = n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
same = n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
$[${EXTEN}=205] ||
$[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
same = n,Hangup()
Now I can send the DTMF from A to B. But How I will get the responce at
server A. I checked all the channels variable but they didn't reply status
of B server channel. All information I will get of server A. Main problem
is that control reach to AGI and then I don't have any rights to do any
update or modification on AGI. So if I can work on request and responce
then it will be the last solution as per my knowledge.
Is this possible with the dialplan or I am just westing time?
On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-12-28 03:25 AM, virendra bhati wrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can
used it?
like A script make the call by using localdail, .call file or any
method.
And after landing the call we send dtmf to IVR system automatically as
per
my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same = n,Read(value,**pleasePress1forSupportPress2fo**
rHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(**suppot,1))
same = n,ExecIf($[${value}=2]?Goto(**help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
We have DTMF based tests for the testsuite[1] that you could use.
[1]
http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/http://svn.asterisk.org/svn/testsuite/asterisk/trunk/
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com http://asterisk.org
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+91-8885268942
Software Engineer
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[asterisk-users] DTMF Testing software to test IVR system
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used it?
like A script make the call by using localdail, .call file or any method.
And after landing the call we send dtmf to IVR system automatically as per
my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same = n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] DTMF Testing software to test IVR system
Hi Satish,
Thank you Satish. I did the same before your e-mail i saw. But i got
another issue in such case.
DTMF is passed to that channels but in case I will make the complete IVR
system for calling server end. and which become so complected to do it.
Is there any alternate way by which I get the response and send DTMF only.
So that complete IVR flow willn't be required to implement at originator
server.
On Wed, Dec 28, 2011 at 2:50 PM, Satish Barot satish4aster...@gmail.comwrote:
Create a callfile with local channel and once first call leg is answered,
use wait() and senddtmf() application on second call leg.
CALLFILE sample:
Channel: LOCAL/1234\@test_ivr
Context: senddtmf
Extension: s
Priority: 1
Extensions.conf sample:
;-- FIRST LEG CALL --;
[test_ivr]
exten = 1234,1,Answer()
same = n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
;--SECOND LEG CALL --;
[senddtmf]
exten = s,1,Noop(# TEST:IVR ##)
; We should wait atleast 'n' of seconds. Where n is length of IVR file in
seconds.
same = n,Wait(10)
same = n,SendDTMF(1)
--SATISH BAROT
On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati virbh...@gmail.comwrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used
it?
like A script make the call by using localdail, .call file or any method.
And after landing the call we send dtmf to IVR system automatically as per
my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same =
n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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Software Engineer
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Re: [asterisk-users] DTMF Testing software to test IVR system
I originate calls from .call file and 1 channel I have at A server A and
another channel at B server.
*A server code is below:-*
exten = 43689956,1,Answer()
same = n,Wait(5)
same = n,SendDTMF(1)
same = n,NoOp(== ${CHANNEL(state)}== state)
same = n,wait(2)
same = n,SendDTMF(123456789012345#)
same = n,NoOp(== ${CHANNEL(state)}== state)
same = n,Hangup()
_ _
| A server | ___DTMF Send_= | B server |
|_| =--- Responce - |_|
*B server code is below:-*
At B server call come to 201 extension which is mention here..
exten = _20[1-6],1,Answer()
same = n,Ringing()
same = n,wait(2)
same = n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
same = n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
$[${EXTEN}=205] ||
$[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
same = n,Hangup()
Now I can send the DTMF from A to B. But How I will get the responce at
server A. I checked all the channels variable but they didn't reply status
of B server channel. All information I will get of server A. Main problem
is that control reach to AGI and then I don't have any rights to do any
update or modification on AGI. So if I can work on request and responce
then it will be the last solution as per my knowledge.
Is this possible with the dialplan or I am just westing time?
On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-12-28 03:25 AM, virendra bhati wrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used
it?
like A script make the call by using localdail, .call file or any method.
And after landing the call we send dtmf to IVR system automatically as per
my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same = n,Read(value,**pleasePress1forSupportPress2fo**
rHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(**suppot,1))
same = n,ExecIf($[${value}=2]?Goto(**help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
We have DTMF based tests for the testsuite[1] that you could use.
[1]
http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/http://svn.asterisk.org/svn/testsuite/asterisk/trunk/
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com http://asterisk.org
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Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] how to stop hacking of my server
Can you give an example how to set these oprion ... On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini ldard...@gmail.com wrote: 2011/12/27 virendra bhati virbh...@gmail.com Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then 1 days. I need your help *CLI log:- * [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for ' 62.141.54.169' - Wrong password [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register
Re: [asterisk-users] how to used SIPp for sip load testing
Hi Sammy, I did the same and start calling. And it's working find but Now I want to the server max capacity with this script then what is the correct process..? On Tue, Dec 27, 2011 at 6:36 PM, Sammy Govind govoi...@gmail.com wrote: Hi, as the Logs say clearly you need to create an extension in default context named service [default] . exten = service,1,NOOP(Incoming call from SIPp) . Regards, Sammy On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati virbh...@gmail.comwrote: Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. haddock8-astrx*CLI -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to stop hacking of my server
Thank you Leandro, Now i am able to register with fix IP. On Tue, Dec 27, 2011 at 3:10 PM, Leandro Dardini ldard...@gmail.com wrote: With deny you'll deny all IP with permit you'll permit only your IP. Yes, it is mandatory to define both deny and permit. Leandro 2011/12/27 virendra bhati virbh...@gmail.com okay, So it is mandatory to define both permit and deny ? if I will update like [trunk1] context=fromoutside type=friend http://0.0.0.0/0.0.0.0 permit=34.2.10.24 qualify=yes So will it be fine or not ? Or it will get rest information from sip.conf general section ? On Tue, Dec 27, 2011 at 2:21 PM, Leandro Dardini ldard...@gmail.comwrote: Yes, this is one of my entries: [trunk1] context=fromoutside type=friend deny=0.0.0.0/0.0.0.0 permit=34.2.10.24 qualify=yes 2011/12/27 virendra bhati virbh...@gmail.com Can you give an example how to set these oprion ... On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini ldard...@gmail.comwrote: 2011/12/27 virendra bhati virbh...@gmail.com Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then 1 days. I need your help -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to used SIPp for sip load testing
Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. haddock8-astrx*CLI -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to stop hacking of my server
okay, So it is mandatory to define both permit and deny ? if I will update like [trunk1] context=fromoutside type=friend http://0.0.0.0/0.0.0.0 permit=34.2.10.24 qualify=yes So will it be fine or not ? Or it will get rest information from sip.conf general section ? On Tue, Dec 27, 2011 at 2:21 PM, Leandro Dardini ldard...@gmail.com wrote: Yes, this is one of my entries: [trunk1] context=fromoutside type=friend deny=0.0.0.0/0.0.0.0 permit=34.2.10.24 qualify=yes 2011/12/27 virendra bhati virbh...@gmail.com Can you give an example how to set these oprion ... On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini ldard...@gmail.comwrote: 2011/12/27 virendra bhati virbh...@gmail.com Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then 1 days. I need your help -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to stop hacking of my server
Yes Eric, I read the archive and found that all guys was saying another open sources project for protection on server like fail2ban. But I want security at configuration level only. As *Leandro* suggest permit and deny option of Sip.conf and *Carlos* suggest the naming process. like that someone suggest that naming should be the SIP phone MAC address. All these are the best for starting security at configuration level. thanks all who posted in this thread. I will used and try Fail2ban but on another server. On Tue, Dec 27, 2011 at 11:19 PM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - Le 27/12/2011 16:04, Tim Nelson a écrit : - Original Message - On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati virbh...@gmail.com wrote: Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? [...] Odd nobody else mentioned it yet, so I'll do it... Check out fail2ban. [...] He said except iptables. fail2ban is iptables related ;-) Ahhh, yes, it would probably have helped if I read the message in it's entirety. :) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to stop hacking of my server
password -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file withAuthenticateApplication
Okay thanks for clear my doubt about Authenticate() function. I know the process of MySQL and ODBC database connection with extensions.conf. On Fri, Dec 23, 2011 at 6:23 PM, bakko asannu...@gmail.com wrote: ** hello, you can't use authenticate for this scenario. You have to create a databse with two fields: extension and password. Then query the database with func_odbc function. There is a spanish article about this: http://www.voztovoice.org/?q=node/478 Regards - Original Message - *From:* virendra bhati virbh...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Friday, December 23, 2011 3:33 AM *Subject:* Re: [asterisk-users] How to use password file withAuthenticateApplication Hi list, I have upgrade my linux version to Asterisk 1.6.2.20. now Authenticate() function is working. But 1 question I want to add this thread.. I have 3 password in my pass.txt file. i want that only sip 2209( just example,) will come with 1234 pass and 2208 with 1235 and rest will come with 1236 password. So how I can make so ? On Tue, Nov 29, 2011 at 8:16 PM, bakko asannu...@gmail.com wrote: I use this system to authenticate my users and work fine. Asterisk: 1.6.2.20 Asterisk user: root Maybe if you active debug on the Asterisk console, you can find the error. Regards - Bakko -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file with AuthenticateApplication
Hi list, I have upgrade my linux version to Asterisk 1.6.2.20. now Authenticate() function is working. But 1 question I want to add this thread.. I have 3 password in my pass.txt file. i want that only sip 2209( just example,) will come with 1234 pass and 2208 with 1235 and rest will come with 1236 password. So how I can make so ? On Tue, Nov 29, 2011 at 8:16 PM, bakko asannu...@gmail.com wrote: I use this system to authenticate my users and work fine. Asterisk: 1.6.2.20 Asterisk user: root Maybe if you active debug on the Asterisk console, you can find the error. Regards - Bakko -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi not installed and application's details is missing in Asterisk
: SpeechDestroy: SpeechLoadGrammar: SpeechProcessingSound: SpeechStart: SpeechUnloadGrammar: StackPop: StartMusicOnHold: Play Music On Hold StopMixMonitor: StopMonitor: Stop monitoring a channel StopMusicOnHold: Stop Playing Music On Hold StopPlayTones: System: TestClient: TestServer: Transfer: TryExec: TrySystem: UnpauseMonitor: Unpause monitoring of a channel UnpauseQueueMember: UserEvent: Verbose: VMAuthenticate: VoiceMail: VoiceMailMain: Wait: WaitExten: WaitForNoise: WaitForRing: WaitForSilence: WaitMusicOnHold: Wait, playing Music On Hold WaitUntil: While: Zapateller: -= 164 Applications Registered =- haddock8-astrx*CLI all information of application and function are missing but working without an issue. Is this problem due to asterisk upgrading. primarily asterisk was installed with rpm (yum install asterisk) and later installed with Asterisk 1.6.2.20.tar.gz -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIfTime days query
Hi , make variable and then put in funtion GotoIf() like set(day=mon|wed|fri) GotoIfTime(*,$day,1,jan?happynewyears,s,1); On Fri, Dec 23, 2011 at 3:03 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using 1.8. Is there a way you can specify staggered days in a single GotoIfTime command e.g. mon|wed|fri? Thanks in Advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIfTime days query
Hi, It will not work... On Fri, Dec 23, 2011 at 3:18 PM, Ishfaq Malik i...@pack-net.co.uk wrote: So pipes can be used as a secondary delimiter? On Fri, 2011-12-23 at 15:08 +0530, virendra bhati wrote: Hi , make variable and then put in funtion GotoIf() like set(day=mon|wed|fri) GotoIfTime(*,$day,1,jan?happynewyears,s,1); On Fri, Dec 23, 2011 at 3:03 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using 1.8. Is there a way you can specify staggered days in a single GotoIfTime command e.g. mon|wed|fri? Thanks in Advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi not installed and application's details is missing in Asterisk
how I can reboot to kernel version 2.6.18-274.12.1.el5 ? Is there any Linux file by which we can change the default kernel version. I have server at different location and can't select from GUI made after reboot machine. On Fri, Dec 23, 2011 at 5:33 PM, Andreas Sikkema h...@ramdyne.nl wrote: [root@haddock8-astrx dahdi-linux-complete-2.5.0.2+2.5.0.2]# make all make -C linux all make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.5.0.2+2.5.0.2/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/dahdi-linux-complete-2.5.0.2+ 2.5.0.2/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/dahdi-linux-complete-2.5.0.2+ 2.5.0.2/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-194.11.1.el5 kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.5.0.2+2.5.0.2/linux' make: *** [all] Error 2 this is the information of installed kernel. [root@haddock8-astrx dahdi-linux-complete-2.5.0.2+2.5.0.2]# rpm -qa|grep kernel kernel-xen-devel-2.6.18-274.12.1.el5 kernel-debug-devel-2.6.18-274.12.1.el5 kernel-debug-2.6.18-274.12.1.el5 kernel-devel-2.6.18-274.12.1.el5 kernel-doc-2.6.18-274.12.1.el5 kernel-2.6.18-274.12.1.el5 kernel-2.6.18-194.11.1.el5 kernel-headers-2.6.18-274.12.1.el5 kernel-xen-2.6.18-274.12.1.el5 You have headers installed for the kernel version 2.6.18-274.12.1.el5 but the DAHDI build is looking for kernel headers for 2.6.18-194.11.1.el5. Either install those kernel headers or reboot to kernel version 2.6.18-274.12.1.el5 and try again. -- Andreas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using shell script output into phoneprov.conf's custom variables
hi tzafrir, How to used #exec in extension.conf ? asterisk.conf is used to enable this option i know.. On Thu, Dec 22, 2011 at 10:28 PM, Olivier oza_4...@yahoo.fr wrote: 2011/12/22, Tzafrir Cohen tzafrir.co...@xorcom.com: On Thu, Dec 22, 2011 at 11:53:16AM +0100, Olivier wrote: As a workaround, is it possible to use include statements in phoneprov.conf ? Both #include and #exec (the latter: if enabled) should work in any Asterisk config file. Great ! At least, this gives me a workaround : I need to value a custom variable to something like 201112221645 (date and time (in hours and minutes). An include and an external script (ran every minute) would just give me that. Maybe an exec, that would be triggered by an appropriate event, would also do it. Thanks for helping. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help_video call not run
Hi in /var/lib/asterisk/sounds/en i store song2_check file(which is video file ,which has audio format MPEG Layer 3) it's MP3 file so use MP3Player() like that exten = _X.,n,MP3Player(song2_check) but 1st you have installed mpg123 On Wed, Dec 21, 2011 at 12:33 PM, amit anand onewaytoconn...@gmail.comwrote: Hi what is the format of the file you are trying to play with exact codec info. On Tue, Dec 20, 2011 at 19:17, Durgesh Mishra durgesh.mis...@rancoretech.com wrote: Hi all In sip.conf i take as [general] videosupport=yes ; then UDPTL will flow to the remote device [phone1] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h264 allow=h263 [phone2] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h261 allow=h263 in extension.conf [employees] exten = 101,1,Dial(SIP/phone1,10) exten = 102,1,Playback(song2_check) in /var/lib/asterisk/sounds/en i store song2_check file(which is video file ,which has audio format MPEG Layer 3) i dial 102 from 101 phone 101(xlite) has following codec support for H623 H623+ check log as [Dec 20 18:38:01] WARNING[10533] file.c: File song2_check does not exist in any format [Dec 20 18:38:01] WARNING[10533] file.c: Unable to open song2_check (format 0x180400 (ilbc|h263|h263p)): No such file or directory phone1 goes just hung up. no vedio play I want to play video file. Plz tell me ,where i am wrong ,and how i can do it. thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Amit Anand +91 9818559898 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why **CONGESTION** not *****NOANSWER****** ?
(Privacy mode, callee rejected the call)) in new stack -- Executing [08723310476@default:10] ExecIf(SIP/77.240.54.13:5063-0856, 0?noop(Privacy mode, callee chose to send caller to torture menu)) in new stack -- Executing [08723310476@default:11] ExecIf(SIP/77.240.54.13:5063-0856, 0?noop(Error parsing Dial command arguments)) in new stack -- Executing [08723310476@default:12] Wait(SIP/77.240.54.13:5063-0856, 9) in new stack -- Auto fallthrough, channel 'SIP/77.240.54.13:5063-0856' status is 'CONGESTION' -- Executing [h@default:1] NoOp(SIP/77.240.54.13:5063-0856, ) in new stack -- Executing [h@default:2] NoOp(SIP/77.240.54.13:5063-0856, 19*) in new stack -- Executing [h@default:3] NoOp(SIP/77.240.54.13:5063-0856, bye Virendra) in new stack -- SIP/voipon-0855 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [1212@default:3] NoOp(SIP/2209-0854, **CONGESTION**) in new stack -- Executing [1212@default:4] ExecIf(SIP/2209-0854, 0?NoOp(Channel unavailable. On SIP, peer may not be registered.)) in new stack -- Executing [1212@default:5] ExecIf(SIP/2209-0854, 0?noop(Busy signal. The dial command reached its number but the number is busy.)) in new stack -- Executing [1212@default:6] ExecIf(SIP/2209-0854, 0?noop(Call is answered. A successful dial. The caller reached the callee.)) in new stack -- Executing [1212@default:7] ExecIf(SIP/2209-0854, 0?noop(No answer. The dial command reached its number, the number rang for too long, then the dial timed out.)) in new stack -- Executing [1212@default:8] ExecIf(SIP/2209-0854, 0?noop(Call is cancelled. The dial command reached its number but the caller hung up before the callee picked up.)) in new stack -- Executing [1212@default:9] ExecIf(SIP/2209-0854, 1?noop(Congestion. This status is usually a sign that the dialled number is not recognised.)) in new stack -- Executing [1212@default:10] ExecIf(SIP/2209-0854, 0?noop(Privacy mode, callee rejected the call)) in new stack -- Executing [1212@default:11] ExecIf(SIP/2209-0854, 0?noop(Privacy mode, callee chose to send caller to torture menu)) in new stack -- Executing [1212@default:12] ExecIf(SIP/2209-0854, 0?noop(Error parsing Dial command arguments)) in new stack -- Executing [1212@default:13] Hangup(SIP/2209-0854, ) in new stack == Spawn extension (default, 1212, 13) exited non-zero on 'SIP/2209-0854' -- Executing [h@default:1] NoOp(SIP/2209-0854, ) in new stack -- Executing [h@default:2] NoOp(SIP/2209-0854, 1*) in new stack -- Executing [h@default:3] NoOp(SIP/2209-0854, bye Virendra) in new stack Even I hangup the call or answer the call and don't pick the call I always get the same responce at asterisk. -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why **CONGESTION** not *****NOANSWER****** ?
Hi Eric,
thanks not getting correct response.
But if default time is 60 then why I will declared ? It's my though and I
don't declared on dial.
On Wed, Dec 21, 2011 at 6:43 PM, Eric Wieling ewiel...@nyigc.com wrote:
-- Got SIP response 480 Temporarily Unavailable back from
10.10.11.203 this is why you are getting congestion instead of NOANSWER.
Fix that and add a timeout to your dial and it should work.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, December 21, 2011 6:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Sam Govind
Subject: [asterisk-users] Why **CONGESTION** not *NOANSWER** ?
Hi List,
I have a DID number which is routed to my production server. Problem is
that when I dial that DID number from my production number then it's gives
DIALSTATUS to CONGESTION if I don't pick the calls. As per the asterisk it
should reply NO ANSWER.
extensions.conf :-
exten = 08723310476,1,Dial(SIP/2218)
same = n,NoOp(**${DIALSTATUS}**)
same = n,ExecIf($['${DIALSTATUS}'='CHANUNAVAIL']?NoOp(Channel
unavailable. On SIP, peer may not be registered.))
same = n,ExecIf($['${DIALSTATUS}'='BUSY']?noop(Busy signal. The
dial command reached its number but the number is busy.))
same = n,ExecIf($['${DIALSTATUS}'='ANSWER']?noop(Call is answered.
A successful dial. The caller reached the callee.))
same = n,ExecIf($['${DIALSTATUS}'='NOANSWER']?noop(No answer. The
dial command reached its number, the number rang for too long, then the
dial timed out.))
same = n,ExecIf($['${DIALSTATUS}'='CANCEL']?noop(Call is
cancelled. The dial command reached its number but the caller hung up
before the callee picked up.))
same = n,ExecIf($['${DIALSTATUS}'='CONGESTION']?noop(Congestion.
This status is usually a sign that the dialled number is not recognised.))
same = n,ExecIf($['${DIALSTATUS}'='DONTCALL']?noop(Privacy mode,
callee rejected the call))
same = n,ExecIf($['${DIALSTATUS}'='TORTURE']?noop(Privacy mode,
callee chose to send caller to torture menu))
same = n,ExecIf($['${DIALSTATUS}'='INVALIDARGS']?noop(Error
parsing Dial command arguments))
same = n,wait(9)
exten = 1212,1,Answer()
same = n,Dial(SIP/08723310476@voipon)
same = n,NoOp(**${DIALSTATUS}**)
same = n,ExecIf($['${DIALSTATUS}'='CHANUNAVAIL']?NoOp(Channel
unavailable. On SIP, peer may not be registered.))
same = n,ExecIf($['${DIALSTATUS}'='BUSY']?noop(Busy signal. The
dial command reached its number but the number is busy.))
same = n,ExecIf($['${DIALSTATUS}'='ANSWER']?noop(Call is answered.
A successful dial. The caller reached the callee.))
same = n,ExecIf($['${DIALSTATUS}'='NOANSWER']?noop(No answer. The
dial command reached its number, the number rang for too long, then the
dial timed out.))
same = n,ExecIf($['${DIALSTATUS}'='CANCEL']?noop(Call is
cancelled. The dial command reached its number but the caller hung up
before the callee picked up.))
same = n,ExecIf($['${DIALSTATUS}'='CONGESTION']?noop(Congestion.
This status is usually a sign that the dialled number is not recognised.))
same = n,ExecIf($['${DIALSTATUS}'='DONTCALL']?noop(Privacy mode,
callee rejected the call))
same = n,ExecIf($['${DIALSTATUS}'='TORTURE']?noop(Privacy mode,
callee chose to send caller to torture menu))
same = n,ExecIf($['${DIALSTATUS}'='INVALIDARGS']?noop(Error
parsing Dial command arguments))
same = n,Hangup()
exten = h,1,NoOp()
same = n,NoOp(${HANGUPCAUSE}*)
same = n,NoOP(bye Virendra)
asterisk cli:-
-- Executing [1212@default:1] Answer(SIP/2209-0854, ) in new
stack
-- Executing [1212@default:2] Dial(SIP/2209-0854, SIP/
08723310476@voipon) in new stack
== Using SIP RTP CoS mark 5
-- Called 08723310476@voipon
== Using SIP RTP CoS mark 5
-- Executing [08723310476@default:1] Dial(SIP/77.240.54.13:5063-0856,
SIP/2218) in new stack
== Using SIP RTP CoS mark 5
-- Called 2218
-- SIP/2218-0857 is ringing
-- SIP/voipon-0855 is making progress passing it to
SIP/2209-0854
-- Got SIP response 480 Temporarily Unavailable back from 10.10.11.203
-- SIP/2218-0857 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [08723310476@default:2] NoOp(SIP/77.240.54.13:5063-0856,
**CONGESTION**) in new stack
-- Executing [08723310476@default:3]
ExecIf(SIP/77.240.54.13:5063-0856,
0?NoOp(Channel unavailable. On SIP, peer may not be registered.)) in new
stack
-- Executing [08723310476@default:4]
ExecIf(SIP/77.240.54.13:5063-0856,
0?noop(Busy signal. The dial command reached its number but the number is
busy.)) in new stack
-- Executing [08723310476@default:5]
ExecIf
Re: [asterisk-users] How to monitor SIP Trunk on production server
Hi Sammy, Actually we have 2 voip trunk at our server 1 of *Voipon* and 2nd of * Gradwell*. When our balance goes down then they don't auto-refill it, I don't know the reason behind it. Ans some time goes down means Call will not go through from VoIP trunk. So want to make a script in AMI / AGI so that I will check the status all the time of these VoIP trunk. In case if someone or both will go down then I will send E-mail / SMS / to all the relevant guys. So that they will check the issue on that case. On Mon, Dec 19, 2011 at 9:41 AM, Sammy Govind govoi...@gmail.com wrote: If you can explain a bit more in detail what you mean by ensuring that trunk is not down? By monitoring a trunks health I assume you are talking about the qualify response time from a trunk. I developed a script for Zabbix monitoring that was executed as a command by Zabbix with a prameter of peer/trunk name to return its qualify time. Once I get a qualify time from asterisk Zabbix plotted the value on its graphs. You can use AMI or asterisk concole command to do somehting like below: #asterisk -rx sip show peer provider-1 | grep qualify Use awk to extract only the numeric value from output of above. Or you can use AMI to fetch sip peer details and parse the value you require. On Sun, Dec 18, 2011 at 10:26 AM, virendra bhati virbh...@gmail.comwrote: Hi List, I have asterisk 1.6.2.20 installed at production server, I have 2 SIP voip trunk for making outgoing and DID for incoming to server. My question is how I can ensure that trunk is not down at production server, So how I can monitor it's automatically by making any scripts? Any hint will be appreciated -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to monitor SIP Trunk on production server
Hi List, I have asterisk 1.6.2.20 installed at production server, I have 2 SIP voip trunk for making outgoing and DID for incoming to server. My question is how I can ensure that trunk is not down at production server, So how I can monitor it's automatically by making any scripts? Any hint will be appreciated -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play audio file for both Caller and Callee in a call
Hi, Plese give a little example of script so that it will be clear. On Thu, Dec 15, 2011 at 11:09 PM, Jim Dickenson dicken...@cfmc.com wrote: You also use AMI to inject audio into the conversation using the ChanSpy application. -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 15, 2011, at 9:23 AM, Danny Nicholas wrote: You can’t per se, but you can call an AGI using stream? ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of * c.savinov...@itntelecom.com *Sent:* Thursday, December 15, 2011 11:22 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Play audio file for both Caller and Callee in a call ** ** Dear Danny: ** ** How can you use Playback in the middle of 2 channels engaged in a conversation? ** ** Thanks C. Savinovich ** ** Original Message Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call From: Danny Nicholas da...@debsinc.com Date: Thu, December 15, 2011 9:31 am To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Playback? What flavor of Asterisk are you using? *From:* *asterisk-users-boun...@lists.digium.com*asterisk-users-boun...@lists.digium.com [*mailto:asterisk-users-boun...@lists.digium.com*asterisk-users-boun...@lists.digium.com ] *On Behalf Of *ISABEL ORDAS ARNAL *Sent:* Thursday, December 15, 2011 10:29 AM *To:* *asterisk-users@lists.digium.com* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Play audio file for both Caller and Callee in a call Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don’t want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! -- Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. *http://www.tid.es/ES/PAGINAS/disclaimer.aspx*http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- -- _ -- Bandwidth and Colocation Provided by *http://www.api-digital.com*http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: *http://www.asterisk.org/hello* http://www.asterisk.org/hello 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which port should be open for asterisk communication
Hi List, Please tell me which ports should be required open for communication with asterisk. like 5060 for sip calls, 4569 for IAX, 10,000 to 20,000.. Apart from these ports what else is required ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which port should be open for asterisk communication
Hi Sammy, Thanks for fastest reply. I to know just for calling time which port's should asterisk need to be open only On Mon, Dec 12, 2011 at 4:03 PM, Sammy Govind govoi...@gmail.com wrote: Hi, That depends on what else your asterisk is doing i.e if an AMI-based code is running then AMI port needs to be open as well. It also depends what other appliactions are running on asterisk-box which require port opening i.e apache or mysql etc. Regards, Sammy On Mon, Dec 12, 2011 at 3:21 PM, virendra bhati virbh...@gmail.comwrote: Hi List, Please tell me which ports should be required open for communication with asterisk. like 5060 for sip calls, 4569 for IAX, 10,000 to 20,000.. Apart from these ports what else is required ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use Hints in asterisk
Hi All, I read about the *Hint* in asterisk. I want to implements into my server for testing purpose. How to use it ? please help me... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Hints in asterisk
Hi All, I did some google and found some documents on that and finally I got some response from asterisk . Below is the CLI output of my google. *haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:IdleWatchers 0 1 hint matching extension 2218 == Using SIP RTP CoS mark 5 -- Executing [222@bhati-test:1] NoOp(SIP/2218-02c3, Call from Gtalk ) in new stack -- Executing [222@bhati-test:2] NoOp(SIP/2218-02c3, Hint for Extension 2218 is ) in new stack -- Executing [222@bhati-test:3] Set(SIP/2218-02c3, CALLERID(name)=From Google Talk) in new stack -- Executing [222@bhati-test:4] Wait(SIP/2218-02c3, 10) in new stack haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:InUse Watchers 0 1 hint matching extension 2218 -- Executing [222@bhati-test:5] Dial(SIP/2218-02c3, SIP/my_sip_phones) in new stack == Using SIP RTP CoS mark 5 [Dec 6 17:39:38] WARNING[6689]: chan_sip.c:5331 create_addr: No such host: my_sip_phones [Dec 6 17:39:38] WARNING[6689]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/2218-02c3' status is 'CHANUNAVAIL' -- Executing [h@bhati-test:1] NoOp(SIP/2218-02c3, hangup the call now) in new stack haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:Idle Watchers 0 1 hint matching extension 2218 * *Is this the right way to use HINT of asterisk ?* On Tue, Dec 6, 2011 at 3:38 PM, virendra bhati virbh...@gmail.com wrote: Hi All, I read about the *Hint* in asterisk. I want to implements into my server for testing purpose. How to use it ? please help me... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Hints in asterisk
Hi All,
Below bold application gives the correct information with asterisk
*HINT*function.
exten = 222,1,NoOp( Call from Gtalk )
*same = n,NoOp(My phone state is currently
${DEVICE_STATE(SIP/2218)})*
same = n,Set(CALLERID(name)=From Google Talk)
same = n,Wait(10)
same = n,Dial(SIP/my_sip_phones)
Spatially thanks for Sammy who give me the way to get success on that way.
On Tue, Dec 6, 2011 at 5:58 PM, Sammy Govind govoi...@gmail.com wrote:
Hello,
AFAIK Hints are used for looking out for a device state before actually
doing anything. Hints can be used from Dial-plan, AMI, or AGI. An example
can be to look for state of a SIP user.
Read these links for better understanding.
http://www.smartvox.co.uk/astfaq_subscribe_hints.htm
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
Regards,
Sammy.
On Tue, Dec 6, 2011 at 5:14 PM, virendra bhati virbh...@gmail.com wrote:
Hi All,
I did some google and found some documents on that and finally I got some
response from asterisk . Below is the CLI output of my google.
*haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:IdleWatchers 0
1 hint matching extension 2218
== Using SIP RTP CoS mark 5
-- Executing [222@bhati-test:1] NoOp(SIP/2218-02c3, Call
from Gtalk ) in new stack
-- Executing [222@bhati-test:2] NoOp(SIP/2218-02c3, Hint for
Extension 2218 is ) in new stack
-- Executing [222@bhati-test:3] Set(SIP/2218-02c3,
CALLERID(name)=From Google Talk) in new stack
-- Executing [222@bhati-test:4] Wait(SIP/2218-02c3, 10) in
new stack
haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:InUse Watchers 0
1 hint matching extension 2218
-- Executing [222@bhati-test:5] Dial(SIP/2218-02c3,
SIP/my_sip_phones) in new stack
== Using SIP RTP CoS mark 5
[Dec 6 17:39:38] WARNING[6689]: chan_sip.c:5331 create_addr: No such
host: my_sip_phones
[Dec 6 17:39:38] WARNING[6689]: app_dial.c:1747 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/2218-02c3' status is
'CHANUNAVAIL'
-- Executing [h@bhati-test:1] NoOp(SIP/2218-02c3, hangup the
call now) in new stack
haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218 State:Idle
Watchers 0
1 hint matching extension 2218
*
*Is this the right way to use HINT of asterisk ?*
On Tue, Dec 6, 2011 at 3:38 PM, virendra bhati virbh...@gmail.comwrote:
Hi All,
I read about the *Hint* in asterisk. I want to implements into my
server for testing purpose. How to use it ? please help me...
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Hints in asterisk
Hi All,
If you used *DEVICE_STATE *function then there is no need to used *HINT* it
work independently.
It's not become to confusion for me how to when to used *HINT *and
when *DEVICE_STATE
?
*
On Tue, Dec 6, 2011 at 6:20 PM, virendra bhati virbh...@gmail.com wrote:
Hi All,
Below bold application gives the correct information with asterisk
*HINT*function.
exten = 222,1,NoOp( Call from Gtalk )
*same = n,NoOp(My phone state is currently
${DEVICE_STATE(SIP/2218)})*
same = n,Set(CALLERID(name)=From Google Talk)
same = n,Wait(10)
same = n,Dial(SIP/my_sip_phones)
Spatially thanks for Sammy who give me the way to get success on that way.
On Tue, Dec 6, 2011 at 5:58 PM, Sammy Govind govoi...@gmail.com wrote:
Hello,
AFAIK Hints are used for looking out for a device state before actually
doing anything. Hints can be used from Dial-plan, AMI, or AGI. An example
can be to look for state of a SIP user.
Read these links for better understanding.
http://www.smartvox.co.uk/astfaq_subscribe_hints.htm
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
Regards,
Sammy.
On Tue, Dec 6, 2011 at 5:14 PM, virendra bhati virbh...@gmail.comwrote:
Hi All,
I did some google and found some documents on that and finally I got
some response from asterisk . Below is the CLI output of my google.
*haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:IdleWatchers 0
1 hint matching extension 2218
== Using SIP RTP CoS mark 5
-- Executing [222@bhati-test:1] NoOp(SIP/2218-02c3, Call
from Gtalk ) in new stack
-- Executing [222@bhati-test:2] NoOp(SIP/2218-02c3, Hint for
Extension 2218 is ) in new stack
-- Executing [222@bhati-test:3] Set(SIP/2218-02c3,
CALLERID(name)=From Google Talk) in new stack
-- Executing [222@bhati-test:4] Wait(SIP/2218-02c3, 10) in
new stack
haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:InUse Watchers 0
1 hint matching extension 2218
-- Executing [222@bhati-test:5] Dial(SIP/2218-02c3,
SIP/my_sip_phones) in new stack
== Using SIP RTP CoS mark 5
[Dec 6 17:39:38] WARNING[6689]: chan_sip.c:5331 create_addr: No such
host: my_sip_phones
[Dec 6 17:39:38] WARNING[6689]: app_dial.c:1747 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/2218-02c3' status is
'CHANUNAVAIL'
-- Executing [h@bhati-test:1] NoOp(SIP/2218-02c3, hangup the
call now) in new stack
haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:Idle
Watchers 0
1 hint matching extension 2218
*
*Is this the right way to use HINT of asterisk ?*
On Tue, Dec 6, 2011 at 3:38 PM, virendra bhati virbh...@gmail.comwrote:
Hi All,
I read about the *Hint* in asterisk. I want to implements into my
server for testing purpose. How to use it ? please help me...
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] video calls not working
Hi all, So how to open JIRA ticket bcoz I don't have any idea of that On Mon, Dec 5, 2011 at 7:43 PM, Danny Nicholas da...@debsinc.com wrote: Not my idea - just what I came across on google - probably should open a JIRA issue so it gets really resolved instead of hit-and-miss patching. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Saturday, December 03, 2011 2:38 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] video calls not working On 11-11-21 10:07 AM, Danny Nicholas wrote: Two items #1 you only need 1 disallow=all in your sip.conf definition #2 you need to patch rtp.c to define 126 as FORMAT_H263 - this is an xlite response to Asterisk starting music-on-hold during the connect pause. The r on the dial command attempts to do a faux ring which xlite interprets as a MOH request, so if you don't want to patch/recompile, just take the r off of Dial. Why are you manually patching asterisk? Have you created an issue in JIRA about this? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best VoIP conferencing phone ?
Hi Faisal, Thanks for reply but I want hardware wase VoIP device. If know please gussed me. From google I fould the list of below devices but I am not sure that these are best for used or have an issue *1)Polycom SoundStation IP 7000 * *Why it's best: *The Polycom SoundStation IP 7000 is the most advanced conference phone from the Polycom SoundStation lineup and leaves little to be desired. With an amazing 20’ 360 radius, the 7000 is perfect for large conference rooms. The new HD voice quality (22 kHz) allows. * * *2) Polycom Voicestation 500* * * *Why it's a best pick: *The Polycom VoiceStation 500 is one of the best conference phones for a wide variety of reasons. The VoiceStation 500 features amazing call quality, 7’ 360 radius, Bluetooth connectivity, wired connection, background noise reduction, and an attractive design. * * *3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S* * * *Why it's a best pick: *With a 360 10’ radius and 8 microphones, everyone is sure to be heard with the Panasonic KX-TS730S. The multiple microphones allows for everyone sitting in on the conference to be heard uniformly without distortion. * * *4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone* * * *Why it's a best pick: *The Cisco 7937G works via VoIP connection, has stunning call clarity, and features a simplistic but expensive design that is easy to use. Cisco is an industry leader in IT communication products, and the 7937G is no different. The 360 design allows everyone to be heard. * * *5)Polycom SoundStation VTX 1000* * * *Why it's a best pick: *The SoundStation VTX 1000 is an incredible conference phone, but it is very pricey and not as good as advertised. The VTX 1000 is designed for large conference rooms and features upgradable software (which is a huge benefit since the cost is so high), 20’ 360 radius. 6)Polycom® SoundStation® IP 5000* 7) GXP2120 6-line Executive HD IP Phone* On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif fai...@vopium.com wrote: I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. ** ** Regards, ** ** Faisal Hanif ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati *Sent:* Wednesday, November 30, 2011 11:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind *Subject:* [asterisk-users] Best VoIP conferencing phone ? ** ** Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best VoIP conferencing phone ?
Thank you for sharing your exp. with me. On Wed, Nov 30, 2011 at 7:34 PM, Darren Wiebe dar...@aleph-com.net wrote: We've been happy with the polycom IP 7000. Darren Wiebe On Nov 30, 2011 1:40 AM, virendra bhati virbh...@gmail.com wrote: Hi Faisal, Thanks for reply but I want hardware wase VoIP device. If know please gussed me. From google I fould the list of below devices but I am not sure that these are best for used or have an issue *1)Polycom SoundStation IP 7000 * *Why it's best: *The Polycom SoundStation IP 7000 is the most advanced conference phone from the Polycom SoundStation lineup and leaves little to be desired. With an amazing 20’ 360 radius, the 7000 is perfect for large conference rooms. The new HD voice quality (22 kHz) allows. * * *2) Polycom Voicestation 500* * * *Why it's a best pick: *The Polycom VoiceStation 500 is one of the best conference phones for a wide variety of reasons. The VoiceStation 500 features amazing call quality, 7’ 360 radius, Bluetooth connectivity, wired connection, background noise reduction, and an attractive design. * * *3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S* * * *Why it's a best pick: *With a 360 10’ radius and 8 microphones, everyone is sure to be heard with the Panasonic KX-TS730S. The multiple microphones allows for everyone sitting in on the conference to be heard uniformly without distortion. * * *4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone* * * *Why it's a best pick: *The Cisco 7937G works via VoIP connection, has stunning call clarity, and features a simplistic but expensive design that is easy to use. Cisco is an industry leader in IT communication products, and the 7937G is no different. The 360 design allows everyone to be heard. * * *5)Polycom SoundStation VTX 1000* * * *Why it's a best pick: *The SoundStation VTX 1000 is an incredible conference phone, but it is very pricey and not as good as advertised. The VTX 1000 is designed for large conference rooms and features upgradable software (which is a huge benefit since the cost is so high), 20’ 360 radius. 6)Polycom® SoundStation® IP 5000* 7) GXP2120 6-line Executive HD IP Phone * On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif fai...@vopium.com wrote: I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. ** ** Regards, ** ** Faisal Hanif ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati *Sent:* Wednesday, November 30, 2011 11:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind *Subject:* [asterisk-users] Best VoIP conferencing phone ? ** ** Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk
[asterisk-users] Best VoIP conferencing phone ?
Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
