[asterisk-users] Sangoma timing device for OpenVZ - Anyone installed it?
Hello, Anyone out there knows the steps to get a Sangoma UT50 or UT51 VoiceTimer USB stick working with an OpenVZ instance of Asterisk? I have Dahdi + UT50 driver installed on mother node running fine but not sure what to do in OpenVZ which has Asterisk installed. Do I have to install Dahdi on OpenVZ? Do I link the Dahdi from mother node to OpenVZ? Sangoma wiki is not clear. 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
[asterisk-users] Is it possible to provision lock Aastra phones?
Hi everyone; Is it possible to provision lock Aastra phones to provider so that no soft or hard reset can unlock them? 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
Re: [asterisk-users] Is it possible to provision lock Aastra phones?
Thanks Patrick. Do the encrypted config files safe guard against hard resets such as Web Recovery mode - aka holding down 1 # sign at startup? My main purpose is to lock the sets due to contract terms so I'd rather not see user steal the phone and break contract without payment. Regards On Sat, Jul 6, 2013 at 7:46 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 07/06/2013 08:15 AM, Bruce B wrote: Hi everyone; Is it possible to provision lock Aastra phones to provider so that no soft or hard reset can unlock them? Iirc you can use encrypted configs using an app called anacrypt and lock them down. The admin guide (3.2.2) has more details in section 2-14, 5-44 - 5-46 and A-187 - A-189. http://www.aastra.com/cps/rde/**aareddownload?file_id=6950-** 16962-_P06_XMLdsproject=**aastramtype=pdfhttp://www.aastra.com/cps/rde/aareddownload?file_id=6950-16962-_P06_XMLdsproject=aastramtype=pdf http://www.aastra.com/**document-library.htm?curr_nav=** 2curr_fam=Aastra+6750iprod_**id=6950#http://www.aastra.com/document-library.htm?curr_nav=2curr_fam=Aastra+6750iprod_id=6950# Regards, Patrick -- __**__**_ -- 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
[asterisk-users] Has iCall gone belly up? iCall carrier services bankrupt?
Hi everyone, Has iCall gone belly or just having really lazy executives / support team? They haven't placed a single long distance call for us since mid last month. Have they run away with deposit money? Are they bankrupt? I appreciate some feedback on this. Thanks, -Bruce -- _ -- 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 check channel status and move on silently?
Hello, I have 10 different routes with few different providers. When I place an international call, I would like the system to try all those routes and place the call through whichever possible. If there is any message but an ANSWER the system should move on to next route. I know this is not the best strategy but there are so many bad routes now-a-days that it's becoming a headache. The only requirement here is to no pass the BUSY or DECLINED codes to end point if that is experienced. I want the user to wait on MOH for example until the call is connected or until all routes are exhausted and then give him a BUSY. What would dialplan for something like this look like in Asterisk 1.8? 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
[asterisk-users] iCall service any good?
Hi everyone, We are getting cotinueous error messages over the past few days from iCall: -- Called iCall/01144 -- Got SIP response 500 Server internal failure back from 72.249.14.242 Is this something everyone else is getting? They are very bad at support and I am not sure if it's their servers or my Asterisk server that is causing the issue. 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
Re: [asterisk-users] Speech recognition in asterisk using google voice API
Hey Zaf, Just checking the Google Speech Recognition package again and I can't see WolframAlpha.agi file. I check all of your projects on Git hub but can't find wolframalpha.agi. Please let us know what the URL is. Thanks, Bruce On Thu, Jan 12, 2012 at 2:49 PM, Lefteris Zafiris zaf@gmail.com wrote: On 01/12/2012 05:50 PM, Danny Nicholas wrote: Two more offerings - #1 - add DTMF parameter so function can be stopped by pressing a digit or digits other than * or # - #2 - add an option to silence the beep. If you were using this in an IVR and wanted to say press 1 or say help for help, silencing the beep before recording would (IMO) make the rendering sound more professional/less mechanical. Both features added: - Usage - agi(speech-recog.agi,[lang],[timeout],[intkey],[NOBEEP]) Records from the current channel untill the timeout (set to 10 seconds by default, -1 for no timeout) is reached or the interrupt key (# by default) is pressed. If NOBEEP is set, no beep sound is played back to the user to indicate the start of the recording. There is now also the option to enable SSL for encrypted communication between your pbx and the google voice server. Updated code can be found here: https://github.com/zaf/asterisk-speech-recog/tarball/master Lefteris Zafiris -- _ -- 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
[asterisk-users] SIP over SSL TCP or SRTP?
Hello, Which one of these ensures that SIP packets are sent and received in a secure format so that users using public wifi don't allow MITM type of attacks or others can't read the plaintext SIP packet info. VPN is not an option. Looking for 2nd most secure to VPN. P.S. Are both options part of the configs of Asterisk or need modules to be selected and installed before doing the configs? 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
Re: [asterisk-users] SIP over SSL TCP or SRTP?
Thanks. Want to secure everything and anything possible. 1- Can both SIP over TLS and SRTP work in conjunction to each other? 2- Is SIP over TLS a package or added on module that can be installed from Digium Asterisk repository? 3- SRTP takes care of the RTP and makes it secure so that MITM type sniffing is not possible? Regards, On Fri, Jun 22, 2012 at 2:39 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 06/22/2012 12:56 PM, Bruce B wrote: Which one of these ensures that SIP packets are sent and received in a secure format so that users using public wifi don't allow MITM type of attacks or others can't read the plaintext SIP packet info. VPN is not an option. Looking for 2nd most secure to VPN. SIP over TLS (what used to be called SSL) is what secures the SIP signaling. SRTP is for securing media streams. -- 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 -- _ -- 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] CDRs do not record in asteriskcdrdb using Digium repository
Yes, asterisk18-addons and asterisk18-addons-mysql are installed. The file cdr_mysql.so exists and I added it to modules.conf with load = cdr_mysql.so. But the module doesn't show loaded when I do module show like cdr. Seems like some config is missing. Which file is responsible for this type of config. Thanks On Sat, Jun 16, 2012 at 4:35 PM, isr...@gmail.com wrote: Did you install the addons Yum install asterisk18-addons-mysql -Original Message- From: Duncan Turnbull dun...@e-simple.co.nz Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 17 Jun 2012 08:30:00 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository -- _ -- 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
Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository
Pinpointed the problem to do with Digium repository. When I do yum install asterisk18 system installs: asterisk18.i386 *1.8.7.0-2_centos5*asterisk-current After, when I do yum update asterisk18-* then the asterisk18-core updates: asterisk18.i386 *1.8.13.0-1_centos5 * * * I don't know if this is a bug in Digium repository or what but 1.8.7.0-2 should NOT show as asterisk-current. Problem is that upon update, not all packages update. So, when trying to do module load cdr_mysql this error prints: *loader.c: Module 'cdr_mysql.so' was not compiled with the same compile-time options as this version of Asterisk.* *loader.c: Module 'cdr_mysql.so' will not be initialized as it may cause instability.* *loader.c: Module 'cdr_mysql' could not be loaded.* * * I tried download .rpm files of asterisk18-addons.rpm, asterisk18-addons-core.rpm, and asterisk18-mysql.rpm but the asterisk18-mysql.rpm fails due to dependencies upon install. So, this seems a bit complicated but it's probably an easy fix if Digium updates the system to use all REAL current version at first install instead of needing to update right after fresh install. Any thoughts? Thanks On Sun, Jun 17, 2012 at 9:34 PM, Bruce B bruceb...@gmail.com wrote: Yes, asterisk18-addons and asterisk18-addons-mysql are installed. The file cdr_mysql.so exists and I added it to modules.conf with load = cdr_mysql.so. But the module doesn't show loaded when I do module show like cdr. Seems like some config is missing. Which file is responsible for this type of config. Thanks On Sat, Jun 16, 2012 at 4:35 PM, isr...@gmail.com wrote: Did you install the addons Yum install asterisk18-addons-mysql -Original Message- From: Duncan Turnbull dun...@e-simple.co.nz Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 17 Jun 2012 08:30:00 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository -- _ -- 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
Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository
This is not related to Asterisk Now but simply Asterisk as provided by Digium repositories and documented in Asterisk Wiki. Source install is one way to do this but that is not the issue in question. I hope someone at Digium fixes and update the repositories to current version. On Sun, Jun 17, 2012 at 10:59 PM, Ikka Vertika ikka.vert...@mitrakreasindo.com wrote: Please correct me if I’m wrong... ** ** The current version of asterisk now is 10.x Cdr_mysql is not used anymore. Now they using odbc to connect to mysql database. ** ** Why dont you try to install asterisk using source TAR.GZ ? It will make you learned where you have to do some setting... :D. Rather difficult but fun... :D ** ** ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* 18 Juni 2012 9:29 *To:* isr...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository ** ** Pinpointed the problem to do with Digium repository. When I do yum install asterisk18 system installs: asterisk18.i386 *1.8.7.0-2_centos5*asterisk-current ** ** After, when I do yum update asterisk18-* then the asterisk18-core updates: asterisk18.i386 *1.8.13.0-1_centos5 * ** ** I don't know if this is a bug in Digium repository or what but 1.8.7.0-2 should NOT show as asterisk-current. ** ** Problem is that upon update, not all packages update. So, when trying to do module load cdr_mysql this error prints: *loader.c: Module 'cdr_mysql.so' was not compiled with the same compile-time options as this version of Asterisk.* *loader.c: Module 'cdr_mysql.so' will not be initialized as it may cause instability.* *loader.c: Module 'cdr_mysql' could not be loaded.* ** ** I tried download .rpm files of asterisk18-addons.rpm, asterisk18-addons-core.rpm, and asterisk18-mysql.rpm but the asterisk18-mysql.rpm fails due to dependencies upon install. So, this seems a bit complicated but it's probably an easy fix if Digium updates the system to use all REAL current version at first install instead of needing to update right after fresh install. ** ** Any thoughts? ** ** Thanks ** ** ** ** ** ** ** ** On Sun, Jun 17, 2012 at 9:34 PM, Bruce B bruceb...@gmail.com wrote: Yes, asterisk18-addons and asterisk18-addons-mysql are installed. The file cdr_mysql.so exists and I added it to modules.conf with load = cdr_mysql.so. But the module doesn't show loaded when I do module show like cdr. ** ** Seems like some config is missing. Which file is responsible for this type of config. ** ** Thanks ** ** ** ** ** ** ** ** On Sat, Jun 16, 2012 at 4:35 PM, isr...@gmail.com wrote: Did you install the addons Yum install asterisk18-addons-mysql -Original Message- From: Duncan Turnbull dun...@e-simple.co.nz Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 17 Jun 2012 08:30:00 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository -- _ -- 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 -- _ -- 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
Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository
Seems like there are new instructions for installing from RPM repository which seems to be working fine and updating to proper current version of Asterisk: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS5%2FRedHatEnterpriseLinux5%29 -Bruce On Sun, Jun 17, 2012 at 11:13 PM, Bruce B bruceb...@gmail.com wrote: This is not related to Asterisk Now but simply Asterisk as provided by Digium repositories and documented in Asterisk Wiki. Source install is one way to do this but that is not the issue in question. I hope someone at Digium fixes and update the repositories to current version. On Sun, Jun 17, 2012 at 10:59 PM, Ikka Vertika ikka.vert...@mitrakreasindo.com wrote: Please correct me if I’m wrong... ** ** The current version of asterisk now is 10.x Cdr_mysql is not used anymore. Now they using odbc to connect to mysql database. ** ** Why dont you try to install asterisk using source TAR.GZ ? It will make you learned where you have to do some setting... :D. Rather difficult but fun... :D ** ** ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* 18 Juni 2012 9:29 *To:* isr...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository ** ** Pinpointed the problem to do with Digium repository. When I do yum install asterisk18 system installs: asterisk18.i386 *1.8.7.0-2_centos5*asterisk-current ** ** After, when I do yum update asterisk18-* then the asterisk18-core updates: asterisk18.i386 *1.8.13.0-1_centos5 * ** ** I don't know if this is a bug in Digium repository or what but 1.8.7.0-2 should NOT show as asterisk-current. ** ** Problem is that upon update, not all packages update. So, when trying to do module load cdr_mysql this error prints: *loader.c: Module 'cdr_mysql.so' was not compiled with the same compile-time options as this version of Asterisk.* *loader.c: Module 'cdr_mysql.so' will not be initialized as it may cause instability.* *loader.c: Module 'cdr_mysql' could not be loaded.* ** ** I tried download .rpm files of asterisk18-addons.rpm, asterisk18-addons-core.rpm, and asterisk18-mysql.rpm but the asterisk18-mysql.rpm fails due to dependencies upon install. So, this seems a bit complicated but it's probably an easy fix if Digium updates the system to use all REAL current version at first install instead of needing to update right after fresh install. ** ** Any thoughts? ** ** Thanks ** ** ** ** ** ** ** ** On Sun, Jun 17, 2012 at 9:34 PM, Bruce B bruceb...@gmail.com wrote: Yes, asterisk18-addons and asterisk18-addons-mysql are installed. The file cdr_mysql.so exists and I added it to modules.conf with load = cdr_mysql.so. But the module doesn't show loaded when I do module show like cdr. ** ** Seems like some config is missing. Which file is responsible for this type of config. ** ** Thanks ** ** ** ** ** ** ** ** On Sat, Jun 16, 2012 at 4:35 PM, isr...@gmail.com wrote: Did you install the addons Yum install asterisk18-addons-mysql -Original Message- From: Duncan Turnbull dun...@e-simple.co.nz Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 17 Jun 2012 08:30:00 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository -- _ -- 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
[asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository
Hello, I have done yum install asterisk18 freepbx and it has installed Asterisk and FreePBX just fine. However, none of the CDR get recorded in asteriskcdrdb table in MySQL. They are available in /var/log/asterisk/cdr-csv/Master.csv. What configuration file sets the setting for writing these CDRs to MySQL? 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
[asterisk-users] Fax .pdf from Asterisk
Hello, I want to send out 1000 faxes. I have an excel sheet of numbers and I have Asterisk 1.8 installed from repository. I don't want to use a fax machine or any ATAs or analogue equipment. How would Asterisk help me with faxing these? and what add-ons do I need to make this possible? I can work my way around doing bash script and do Asterisk spool files, but I am unclear as to what happens from that point on to getting the result of fax sent or not. Some guidance is much appreciated. Thanks, Bruce -- _ -- 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] Fax .pdf from Asterisk
Lee, Much appreciated for the input. I am running all VoIP. SIP and IAX2 to our ITSPs. Please elaborate on HylaFax and IAXmodems. Is there a guide posted on to get it running, or is it part of the repository? Once installed how would one send .pdf as fax? Thanks, Bruce On Thu, May 3, 2012 at 7:42 PM, Lee Howard fax...@howardsilvan.com wrote: On 05/03/2012 01:28 PM, Bruce B wrote: I want to send out 1000 faxes. I have an excel sheet of numbers and I have Asterisk 1.8 installed from repository. I don't want to use a fax machine or any ATAs or analogue equipment. How would Asterisk help me with faxing these? and what add-ons do I need to make this possible? Not interested in HylaFAX with IAXmodems? (I presume that you are using PSTN circuits and not VoIP.) Thanks, Lee. -- _ -- 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] Fax .pdf from Asterisk
James,
That is amazing details. I can use all of this. Thank you for sharing.
I am assuming you installed res_fax from repository?
*yum install asterisk18-res_fax_digium.i386*
And how did you install SpanDSP? Is there a guide you used?
I am aiming for multi-channels fax so the digium one won't do for me as
it's one channel limit like you mentioned. I probably don't need T.38 but
hey it won't hurt to have it.
Thanks again,
On Fri, May 4, 2012 at 12:37 AM, James Sharp ja...@fivecats.org wrote:
On 5/3/12 9:16 PM, Bruce B wrote:
Lee,
Much appreciated for the input.
I am running all VoIP. SIP and IAX2 to our ITSPs. Please elaborate on
HylaFax and IAXmodems. Is there a guide posted on to get it running, or
is it part of the repository? Once installed how would one send .pdf as
fax?
You can either use the Free Fax for Asterisk single channel at a time
fax system or app_fax that uses the SpanDSP library. I'm working on a
similar system myself that uses the SpanDSP system because I could never
get FFFA to talk T.38 right to my provider (Gafachi).
I use spoolfiles to create a call that lands in my dialplan and from
there, I can pick up the fax results in the dialplan.
Here's my callfile
Channel: SIP/1771655@gafachi1a
CallerID: 18005551212
MaxRetries: 0
RetryTime: 60
WaitTime: 30
Context: faxout
Extension: 5
Priority: 1
And my extensions file
[faxout]
exten = 5,1,SendFAX(/tmp/test.tiff,a)
exten = 5,n,Noop(${LOCALSTATIONID});
exten = 5,n,Noop(${LOCALHEADERINFO});
exten = 5,n,Noop(${FAXSTATUS});
exten = 5,n,Noop(${FAXERROR});
exten = 5,n,Noop(${REMOTESTATIONID});
exten = 5,n,Noop(${FAXPAGES});
exten = 5,n,Noop(${FAXBITRATE});
exten = 5,n,Noop(${FAXRESOLUTION});
Note that I use the a option in SendFAX. That makes it change behaviors
on negotiating things like T.38. It was needed for my provider, but it may
not be needed for yours. You may or may not need/even have access to T.38
faxing.
You will have to convert the PDF to a TIFF file before you can send it
with either Fax subsystem.
gs -q -dNOPAUSE -dBATCH -sDEVICE=tiffg4 -sPAPERSIZE=letter
-sOutputFile=/tmp/test.tiff test.pdf
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Re: [asterisk-users] Best CRM for Asterisk
I am looking for the same thing as Virendra, Easy to deploy open source. Vicidial and Goautodial are hard to deploy and too blotted. Vicidial is also ugly interface. No diss and I know that it's the best out there but not the easy to deploy. Goautodial people didn't even show interest configuring it for us when we asked to pay them so that's out of the window for us too. Any other suggestions? Best, On Fri, Feb 24, 2012 at 1:42 PM, mahesh katta maheshka...@flexydial.comwrote: The best call center solution is vicidialnow, now Goautodial both are same. Contact with Buzzworks Business Pvt.Ltd, those are give good Callcenter solution will give you. Best Regards, Mahesh Katta On Sat, Feb 25, 2012 at 12:02 AM, virendra bhati virbh...@gmail.comwrote: Hi All, I want to know the best CRM which will work with asterisk and also is Open Source. vTiger, AsterCRM,SugarCRM etc. Which is easy to deploy and have all feature of Call Center. Please help me I never did work for call center solution. 1st time i am trying to make it. I have knowledge of asterisk. and make calling card solution with a2billing. this is the new task for me. Give me your suggestion. thanks in advance.. -- 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 -- _ -- 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
virbh...@gmail.com wrote: how many UDP ports is required for 1 call. and why . If you mean a voice call, it appears that each host must open three UDP sockets: - One to send/receive SIP commands - Two to receive sound (one for RTP, one for RTCP; The first port is even, the other is odd) This is I think the best answer provided to you so far. The simplest and most relevant I should say. Just to add to that, first port is TCP port and the other two are UDP ports assuming you are using SIP protocol. -Bruce -- _ -- 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] Should you ever use nat=no?
On Thu, Feb 16, 2012 at 12:30 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/11/2012 06:59 PM, Bruce B wrote: If your server is open to the internet and in SIP general section you have nat=no and in peers you have nat=yes or vice versa then it's possible to enumerate your extension. Because Asterisk responds with different messages if the extension exists or not based on that difference in the nat setting then it's possible to tell if an extension 100 exists or not. Over the past few years, Digium has come to realization to respond to all unauthenticated calls the same way in order to thwart any attack attempts or guesses on the extension but it's still not perfect yet as these improvements are done at a really slow pace. Regardless, they are being made and there truely is a security risk. really slow pace? Please point out any one of these issues that took an unnecessarily long time to resolve once it was identified and brought to the development team's attention. Was referring to general state and mindset of logging and standardizing it for security tools. I was not referring to any Jira issues particularly though sometimes it takes a lot to convince the dev team something is a bug or missing. Security should be taken more importantly and I don't feel it is. A good example is that of core set verbose 0 effecting all the logging and rendering all security tools useless. There is another thread going on about this right now... I always use nat=yes. I don't even know why nat=no exists as there is nothing that can't be done with nat=yes. Plus nat=yes will take care of some of the surprise one-way audio scenarios as well so why use nat=no at all?! I vote to totally get rid of the nat setting all together and hard code it and set it to yes but again there are others who may not agree. As was already pointed out in the discussions that lead up to the 'nat' default being changed, there are SIP endpoints out there that do not work properly with 'nat=yes' (or 'nat=force_rport'). They behave improperly when Asterisk adds an 'rport' parameter to the top-level Via header in its responses. Setting 'nat=no' is the only way to keep this from happening. I agree. Wish they followed a standard to make everyone's life easier. -- 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 -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
Wouldn't a shell script be a band-aid solution? CLI verbose should have absolutely no effect on other loggings. I have been saying this forever that Asterisk logging should be very strong and separate of anything else including what we see on the CLI. This is important for security reasons. You forget to put the verbose back to 9 then your Fail2ban stop working. You are debugging the server and playing with core set verbose then you are momentarily opening for attacks. I do understand what core set verbose was initially made for but these things are not set in stone and should be improved given security is becoming such a huge issue. Separating logger.conf from core set verbose is the best solution. Best, On Thu, Feb 16, 2012 at 10:11 PM, Luke Hamburg l...@solvent-llc.com wrote: Fair enough. Giving up on the backport to 1.8 or 10 for now, I had a thought for a kludge. How about a shell script (scheduled with cron) that checks for any 'active' consoles -- any connected consoles where there has been user input within the last X minutes. If it finds none, then set the verbosity back to 5 (or whatever level you want). There are a few problems with this -- I couldn't find any way to: 1) query Asterisk for a count or list of console connections, much less 'active' ones 2) query Asterisk for the current verbosity level (without changing it) Am I barking up another wrong tree here? Anyone have any other ideas on how to solve this problem? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Thursday, February 16, 2012 8:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] High verbose set at console effects the logger file Full - Why is that? It's not in Asterisk 10, it's in the current trunk, which will eventually become Asterisk 11. The patch, while a very nice and useful enhancement, is unfortunately fairly intrusive. I can't see it becoming part of the Asterisk 1.8 or Asterisk 10 branches, given (a) the fact that it is certainly an improvement and not a bug fix, and (b) the risk involved in back-porting a patch of that magnitude and scope. Matthew Jordan Digium, Inc. | Software Developer -- _ -- 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
Re: [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?
Sammy, Would you care to elaborate please. Have you had experience doing such a campaign using AMI? Maybe you can share of the code. Most appreciated, On Sat, Feb 11, 2012 at 10:15 AM, Sammy Govind govoi...@gmail.com wrote: I'd definitely go with AMI ! On Sat, Feb 11, 2012 at 6:39 PM, asterisk jobs asteriskcod...@gmail.comwrote: Thanks for the input but using spool files or AMI or AGI is way different from each other and that is what I want to get an input on. I do have hands on with all methods like I noted but want to know what the trend is now-a-days and what is more robust and proven out of all three. Best, On Sat, Feb 11, 2012 at 8:12 AM, David Backeberg dbackeb...@gmail.comwrote: On Sat, Feb 11, 2012 at 8:03 AM, asterisk jobs asteriskcod...@gmail.com wrote: Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the easiest way to approach the issue. The way to call 5000 numbers is to call one number, really well. Then you put it in a loop. You need to run a lab for long enough that you have the bugs worked out, before you subject real people to problems. With asterisk you can always tell the real-time status of a call, even if you initiate from a call file. Perhaps you would enjoy reading up on Local channels. Some people prefer to initiate calls from AMI. I tried it and didn't like it. But because most of us have been annoyed by an autodialer in our lives, even if we ourselves have made autodialers in the past, this is probably about the limit of the help you're going to get, unless you ask a more specific question that shows you've been trying to learn this hands-on and you've gotten stuck on a particular 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-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 -- _ -- 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] Should you ever use nat=no?
If your server is open to the internet and in SIP general section you have nat=no and in peers you have nat=yes or vice versa then it's possible to enumerate your extension. Because Asterisk responds with different messages if the extension exists or not based on that difference in the nat setting then it's possible to tell if an extension 100 exists or not. Over the past few years, Digium has come to realization to respond to all unauthenticated calls the same way in order to thwart any attack attempts or guesses on the extension but it's still not perfect yet as these improvements are done at a really slow pace. Regardless, they are being made and there truely is a security risk. I always use nat=yes. I don't even know why nat=no exists as there is nothing that can't be done with nat=yes. Plus nat=yes will take care of some of the surprise one-way audio scenarios as well so why use nat=no at all?! I vote to totally get rid of the nat setting all together and hard code it and set it to yes but again there are others who may not agree. -Bruce On Sat, Feb 11, 2012 at 6:54 PM, sean darcy seandar...@gmail.com wrote: I've been lurking on the dev discussion on creating nat=auto. It all leads me to think there's no reason to use nat=no. We have about 60 internal sip extensions connected to an multihomed asterisk box where the external ip is not nat'ed. Each of the internal sip contexts has nat=no. On startup I get a slew of warnings about intruders being able to distinguish real extensions. But that isn't right, is it? Or if it is, wouldn't the intruder have to be on the inside 10.0.0.0 net? But so what? Does nat=no buy you anything? faster? slicker? richer? sean -- __**__**_ -- 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
Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.
On Fri, Jan 6, 2012 at 8:16 AM, Eric Wieling ewiel...@nyigc.com wrote: All screwing up with Asterisk is supposed to be documented in the relevant UPGRADE*.txt files. Have you checked them? is supposed to be but does NOT happen. There are many examples of regressions introduced after many complains on the forum of a broken feature. -- _ -- 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] Speech recognition in asterisk using google voice API
Does sox have more features on a Debian system than RHEL? Is that why it won't work on RHEL? Cheers, On Wed, Jan 4, 2012 at 6:42 PM, Lefteris Zafiris zaf@gmail.com wrote: Fresh code is out! The use of sox can be now optionally enabled by the user if the system has a recent version of the program (won't work in RHEL/Centos 5) This is done by editing the script and setting the variable 'use_sox'. When sox is used the audio gets normalized, low frequency noise (100Hz) is removed and also possible DC offset is corrected. Those are supposed to improve the recognition results(?). The settings are still a bit experimental, feel free to play with them and report what settings improved your results. get the new version here: https://github.com/downloads/zaf/asterisk-speech-recog/asterisk-speech-recog-0.3.tar.gz Lefteris Zafiris -- _ -- 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
Re: [asterisk-users] Speech recognition in asterisk using google voice API
Thanks. I have been testing Aastra phones with SIP and had great results. I am testing my cell phone now and sometimes get -1 for id, status, utterance, and confidence. What does that mean? Cheers On Fri, Jan 6, 2012 at 9:40 PM, Lefteris Zafiris zaf@gmail.com wrote: On Fri, 6 Jan 2012 20:46:14 -0500 Bruce B bruceb...@gmail.com wrote: Does sox have more features on a Debian system than RHEL? Is that why it won't work on RHEL? RHEL's 5 version of sox is really old and outdated. The command syntax and the switches are totally different compared to recent versions of sox. Anyway I'm not sure audio normalization and the rest we use sox for is really needed. My tests so far didn't show any improvements in detection rates. Keep in mind that all this is still WIP and the option to use sox is more for testing than for serious use. Lefteris Zafiris -- _ -- 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
Re: [asterisk-users] Speech recognition in asterisk using google voice API
NVM. I explored the code and see the logic. I had sox = 1 so it was failing on RHEL. To report, my cell phone from a PRI gets same confidence level just like SIP. Building my control app now. Should make my life much easier while driving. Thanks again :-) -Bruce On Fri, Jan 6, 2012 at 10:50 PM, Bruce B bruceb...@gmail.com wrote: Thanks. I have been testing Aastra phones with SIP and had great results. I am testing my cell phone now and sometimes get -1 for id, status, utterance, and confidence. What does that mean? Cheers On Fri, Jan 6, 2012 at 9:40 PM, Lefteris Zafiris zaf@gmail.comwrote: On Fri, 6 Jan 2012 20:46:14 -0500 Bruce B bruceb...@gmail.com wrote: Does sox have more features on a Debian system than RHEL? Is that why it won't work on RHEL? RHEL's 5 version of sox is really old and outdated. The command syntax and the switches are totally different compared to recent versions of sox. Anyway I'm not sure audio normalization and the rest we use sox for is really needed. My tests so far didn't show any improvements in detection rates. Keep in mind that all this is still WIP and the option to use sox is more for testing than for serious use. Lefteris Zafiris -- _ -- 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
Re: [asterisk-users] Speech recognition in asterisk using google voice API
Added two new features to the script: Timeout value and speechdata type. *exten = s,n,agi(speech-recog.agi,en-US,3000,phoneNumb)* - Will listen for 3 seconds and sanitize return as a single number without any spaces in between. This helps when one reads phone number in format 415-554-2323 and google returns, 415 554 2323 as result which is not very usable. *exten = s,n,agi(speech-recog.agi,en-US,2,string)* - Will listen for 20 second and return result as provided by Google untouched. It would be great to see them in future versions as I seem to need them dearly in a real life scenario. Updated script attached. -Bruce On Fri, Jan 6, 2012 at 11:03 PM, Bruce B bruceb...@gmail.com wrote: NVM. I explored the code and see the logic. I had sox = 1 so it was failing on RHEL. To report, my cell phone from a PRI gets same confidence level just like SIP. Building my control app now. Should make my life much easier while driving. Thanks again :-) -Bruce On Fri, Jan 6, 2012 at 10:50 PM, Bruce B bruceb...@gmail.com wrote: Thanks. I have been testing Aastra phones with SIP and had great results. I am testing my cell phone now and sometimes get -1 for id, status, utterance, and confidence. What does that mean? Cheers On Fri, Jan 6, 2012 at 9:40 PM, Lefteris Zafiris zaf@gmail.comwrote: On Fri, 6 Jan 2012 20:46:14 -0500 Bruce B bruceb...@gmail.com wrote: Does sox have more features on a Debian system than RHEL? Is that why it won't work on RHEL? RHEL's 5 version of sox is really old and outdated. The command syntax and the switches are totally different compared to recent versions of sox. Anyway I'm not sure audio normalization and the rest we use sox for is really needed. My tests so far didn't show any improvements in detection rates. Keep in mind that all this is still WIP and the option to use sox is more for testing than for serious use. Lefteris Zafiris -- _ -- 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 speech-recog.agi Description: Binary data -- _ -- 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 1.8.8 - caller ID not working.
but not it is not working again. I wish they stop screwing up with that Asterisk, they keep introducing new version and more bugs :-/ Wish not granted !!! :-) You will be the guinea pig to new features !!! Same issue with A2Billing connecting to Asterisk. With older version this problem is not there. -Bruce -- _ -- 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] Speech recognition in asterisk using google voice API
Note to self: Never release anything asterisk related without testing on RHEL/Centos 5 Thank you for reporting this. I have replaced sox with flac and it seems to work now on older platforms too (tested on Centos 5 with asterisk 1.4). You can get the updated code here: https://github.com/zaf/asterisk-speech-recog/tarball/master Lefteris Zafiris Works beautifully. Amazing job Lefteris. Thanks. The best result I got in probability was 0.9725632 by saying, hello. I think there is some non-phonetic logic built-in as well. I tried, 1, 2 and I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got, 0.97256315. Probably Google sees the pattern?! What are some of the other tricks (if any) or consideration that one should make while creating a strong speech recognition enabled IVR? Best, -- _ -- 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] Speech recognition in asterisk using google voice API
Very interesting. I just tried to get it to work but it complains about sox. Probably you used a different version of sox? *PBX-*CLI /usr/bin/sox: invalid option -- -* */usr/bin/sox: invalid option -- n* */usr/bin/sox: invalid option -- o* */usr/bin/sox: -r must be given a positive integer* * -- speech-recog.agi: /usr/bin/sox failed: 512* I am using: *Package sox-12.18.1-1.el5_5.1.i386 * Thanks, On Tue, Jan 3, 2012 at 9:42 PM, Lefteris Zafiris zaf@gmail.com wrote: Hello, I have written an agi script that uses google voice API for voice recognition. The script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable. More info and dialplan examples can be found in the README file: https://raw.github.com/zaf/asterisk-speech-recog/master/README The script is available here: https://github.com/zaf/asterisk-speech-recog The code is still young and not roughly tested so comments, suggestions and bug reports are more than welcome. Lefteris Zafiris -- _ -- 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
Re: [asterisk-users] Speech recognition in asterisk using google voice API
And with recent version 14.3.2 I get: /usr/local/bin/sox FAIL formats: no handler for file extension `flac' -- speech-recog.agi: /usr/local/bin/sox failed: 512 -- SIP/-002eAGI Script speech-recog.agi completed, returning 0 Regards, On Wed, Jan 4, 2012 at 12:43 AM, Bruce B bruceb...@gmail.com wrote: Very interesting. I just tried to get it to work but it complains about sox. Probably you used a different version of sox? *PBX-*CLI /usr/bin/sox: invalid option -- -* */usr/bin/sox: invalid option -- n* */usr/bin/sox: invalid option -- o* */usr/bin/sox: -r must be given a positive integer* * -- speech-recog.agi: /usr/bin/sox failed: 512* I am using: *Package sox-12.18.1-1.el5_5.1.i386 * Thanks, On Tue, Jan 3, 2012 at 9:42 PM, Lefteris Zafiris zaf@gmail.comwrote: Hello, I have written an agi script that uses google voice API for voice recognition. The script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable. More info and dialplan examples can be found in the README file: https://raw.github.com/zaf/asterisk-speech-recog/master/README The script is available here: https://github.com/zaf/asterisk-speech-recog The code is still young and not roughly tested so comments, suggestions and bug reports are more than welcome. Lefteris Zafiris -- _ -- 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
[asterisk-users] Wired attack on Asterisk - Can anyone explain this?
Hello, Can anyone explain what this attack was trying to do? *19.19.19.19 *is my server IP and it seems that they are trying to use my server IP to initiate a SIP call to 199.16.208.29 or 199.16.208.30. Is that so? *Call Date Channel Source CLID DST Disposition Duration * 10/4/2011 19:58 SIP/19,19,19,19-061f 111222 199.16.208.29 199.16.208.30 111222 s ANSWERED 3 10/4/2011 19:58 SIP/19,19,19,19-061f 111222 199.16.208.29 199.16.208.30 111222 s ANSWERED 3 10/4/2011 19:58 SIP/19,19,19,19-0620 101 199.16.208.29 199.16.208.30 101 s ANSWERED 2 10/4/2011 19:58 SIP/19,19,19,19-0620 101 199.16.208.29 199.16.208.30 101 s ANSWERED 2 10/4/2011 19:58 SIP/19,19,19,19-0621 1001 199.16.208.29 199.16.208.30 1001 s ANSWERED 1 10/4/2011 19:58 SIP/19,19,19,19-0621 1001 199.16.208.29 199.16.208.30 1001 s ANSWERED 1 10/4/2011 19:59 SIP/19,19,19,19-0622 200 199.16.208.29 199.16.208.30 200 s ANSWERED 1 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
Re: [asterisk-users] High verbose set at console effects the logger file Full - Why is that?
On Sat, Dec 31, 2011 at 5:06 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Fri, Dec 30, 2011 at 09:03:34PM -0500, Bruce B wrote: So, based on what you are saying if I issue the command core set verbose 0 and then exit the system Fail2Ban will stop working for Asterisk (this is since Fail2ban works based on the log file entries). Can anyone else please confirm that as well. Though in trunk you can set different log levels to different files. Tzafrir, thanks for the feedback. Can you please elaborate on that. Is that something that is not effected by the CLI commands? Not sure which trunk you are pointing too. Regards, -- _ -- 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 fail2ban filters - show us yours
Had one of my systems hit this morning too. Asterisk 1.8 branch+FreePBX 2.9 no anonymous. 260 call attemps in 2 minutes. Here is part of the logs. I am updating my filter to see if it helps, THANKS Bruce!!! I am trying to get this working for FreePBX as I think they are more vulnerable than the vanilla Asterisk setups. Some of the charecters might have to be escaped and I am not an expert in Python but trying to learn it so I will post back my findings. In the meanwhile it would be great if others share their findings as well. -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
Hi everyone, I am playing around with Asterisk 1.8.8.0 from Digium repository. This is all there is to my logger.conf file: *[general]* *dateformat=%F %T* * * *[logfiles]* *full = notice,warning,error,debug,verbose,dtmf,fax* * * However, when I do, core set verbose 0 at CLI, Asterisk ceases to write to /var/log/asterisk/full file for some reason. When I type core set verbose 9 at CLI then it starts writing to /var/log/asterisk/full. Is this the correct behaviour or am I missing a config setting? Of course I want the /var/log/asterisk/full file to always keep the logs regardless of what the verbosity at CLI level is. 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
Re: [asterisk-users] High verbose set at console effects the logger file Full - Why is that?
Okay, but I thought that the line console = is supposed to be for CLI and the line Full = is supposed to be for the file /var/log/asterisk/full. Why would the Full = be effected by core set verbose 0? Is this just bad assumption on the part of the developers? I would only assume that core set verbose 0 should only effect what I see at CLI level and not at my my /var/log/asterisk/full log file. Am I missing something? Thanks for the feedback. On Fri, Dec 30, 2011 at 6:19 PM, Jim Dickenson dicken...@cfmc.com wrote: If you want to stop stuff from going to the console you can use the command logger mute and console will not get output but log file will. -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 3:11 PM, Bruce B wrote: Hi everyone, I am playing around with Asterisk 1.8.8.0 from Digium repository. This is all there is to my logger.conf file: *[general]* *dateformat=%F %T* * * *[logfiles]* *full = notice,warning,error,debug,verbose,dtmf,fax* * * However, when I do, core set verbose 0 at CLI, Asterisk ceases to write to /var/log/asterisk/full file for some reason. When I type core set verbose 9 at CLI then it starts writing to /var/log/asterisk/full. Is this the correct behaviour or am I missing a config setting? Of course I want the /var/log/asterisk/full file to always keep the logs regardless of what the verbosity at CLI level is. 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 -- _ -- 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
Re: [asterisk-users] High verbose set at console effects the logger file Full - Why is that?
One can set the verbose level as well as the debug level. These control how much log information is generated at all not where it is being written. What do you mean by above? Can I see something in the logger.conf that will keep it always at certain verbose level regardless of what command I issue at CLI? You see the problem I have is that Fail2ban reads the asterisk full log file. So, if I am playing on the CLI and then do core set verbose 0 and exit the box and forget to set it back to 9 then Fail2ban stops working because the log file hasn't logged the attack. I still think there is a way around this and I am missing a config. Why would anyone tie security logs to a mere CLI command? Thanks again -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
So, based on what you are saying if I issue the command core set verbose 0 and then exit the system Fail2Ban will stop working for Asterisk (this is since Fail2ban works based on the log file entries). Can anyone else please confirm that as well. Thanks again for your input. On Fri, Dec 30, 2011 at 8:36 PM, Jim Dickenson dicken...@cfmc.com wrote: -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 4:55 PM, Bruce B wrote: One can set the verbose level as well as the debug level. These control how much log information is generated at all not where it is being written. What do you mean by above? Can I see something in the logger.conf that will keep it always at certain verbose level regardless of what command I issue at CLI? No the verbose command controls how much verbose stuff is output. The debug command controls how much debug stuff is output. These are absolute controls of that information. As I said in my original email you can turn off stuff going to the CLI with the logger mute command. That way you do not adjust the verbose level at all. You see the problem I have is that Fail2ban reads the asterisk full log file. So, if I am playing on the CLI and then do core set verbose 0 and exit the box and forget to set it back to 9 then Fail2ban stops working because the log file hasn't logged the attack. I still think there is a way around this and I am missing a config. Why would anyone tie security logs to a mere CLI command? Thanks again -- _ -- 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
Re: [asterisk-users] 1.6 and 1.8
Log are being filled with g729 transcoding error in 1.8.7x now :-( I don't dare to test 1.8.8x as it might have something else broken. Unfortunately I can no longer trust the release candidates. Thanks for the input. On Thu, Dec 29, 2011 at 8:29 AM, Ryan Wagoner rswago...@gmail.com wrote: On Thu, Dec 29, 2011 at 12:05 AM, Bruce B bruceb...@gmail.com wrote: I have been running 1.8.7 with a few fixes back ported from the 1.8.8 release candidate for the last 2.5 months. The system processes around 4,000 calls per day over PRIs for 250 Polycom phones. Previously I was running 1.6.1.18 with a bunch of back ports for fixes and features. Overall it was stable but every few months I had an issue where a channel would get hung. When this happened core show channels would crash the console and I would eventually have to restart Asterisk. Ryan What od you mean by, been running 1.8.7 with a few fixes back ported from the 1.8.8 release candidate. So, this is a version 1.8.7 release that you are using or a 1.8.8 or is this a mix of both that you come up with? Can you please be specific with fixes? Thanks It was a mix I came up with as I was hitting a few bugs in 1.8.7 and 1.8.8 wasn't released. At this point I would just go for 1.8.8. The issue was mainly 17541 which was filling my logs and basically made Asterisk unusable. https://issues.asterisk.org/jira/browse/ASTERISK-17541 https://issues.asterisk.org/jira/browse/ASTERISK-18570 https://issues.asterisk.org/jira/browse/ASTERISK-18101 I had tested 1.8.4 before and was hit by a bunch of dtmf issues that were fixed in 1.8.5. When 1.8.7 came out it looked fairly stable so I switched from 1.6.1.18. I was running the 1.6.1 branch as I needed TCP SIP support. Right now I have been testing 1.8.8 which looks to be a good release. The 1.8 series has come a long way in a few releases as far as fixing major bugs. Ryan -- _ -- 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
Re: [asterisk-users] Interesting attack tonight fail2ban them
Maybe your logger is not setup properly?! You should get the IP in logs. I can't think of when you won't get the IP in your logs unless the SIP packets are manipulated. That IP is from Voxel.net. You don't have a VPS or service from them do you? 2011/12/29 Michelle Dupuis mdup...@ocg.ca 1. I checked the log and I don't see any registration attempt, so I *assume* they simply send an invite, and so they are in the external/outside context of my dialplan. So they are trying to reach extensions which don't exist. If they succesfully registered they would be on the internal context, and their calls would have succeeded. (Or am I missing something?). I actually see nothing in the log but the notice (and nothing on the CLI but the notice)...so I assume it is only an invite? 2. I got their IP by turning on SIP DEBUG while they were attacking. 3. The NOTICE showed a call from '' - what normally goes there? I can't reproduce this NOTICE so I'm not sure what causes it to be recorded. Normal calls show Accepting AUTHENTICATED call from x.x.x.x I'm thinking of using SIPCHANINFO and LOG to log the bad attempts, and let fail2ban takeover from there. Thanks -- *From:* asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Mikhail Lischuk [ mlisc...@itx.com.ua] *Sent:* Thursday, December 29, 2011 4:14 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Interesting attack tonight fail2ban them Jeroen Eeuwes писал 29.12.2011 07:29: Probably my understanding is limited, but it seems to me that they have already 'access' to your Asterisk for them to be able to try to make outgoing calls. Wouldn't it be better to make sure they get the usual errors like Registration from failed - no matching peer found? In other words, how did they get this far in the first place? Best regards, Jeroen Eeuwes Agreed. If you didn't get the Failed to authenticate on INVITE (or whatever error should Asterisk log for not authenticated user trying to place a call, I might be wrong here) - your problem is way more serious. As I can advice you from my wast (despite not always successfull) intruders fighting experience - banning by useragent can help. I always dreamed of Asterisk to implement that, but until then - if all your users are like Linksys blablabla or eyeBeam blablabla and you see any other agent on the Asterisk log - just ban it. Ofcourse, there are 2 limitations: 1) If he doesnt register, Asterisk wont show his useragent in log. And as for yor issue - neither will it show IP. I think we might ask devs to correct that some day 2) if you dont have some standard for user sip devices and they use whatever they want to, it wont help either -- With Best Regards Mikhail Lischuk mlisc...@itx.com.ua ITX Ukraine -- _ -- 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
Re: [asterisk-users] Asterisk fail2ban filters - show us yours
Hi,
I Have added this line for asterisk 1.8 (i have allowguest=yes and
context=default in sip.conf):
NOTICE.* .*: Call from '.*' (HOST) to extension '.*' rejected because
extension not found in context 'default'.
Em 29-12-2011 13:03, Patrick Lists escreveu:
Hi,
In the thread Interesting attack tonight fail2ban them Bruce B
mentioned it would be nice to have input from the Community to come up with
the best set of fail2ban filters. That's a great idea. So let's start with
Bruce's filters (thanks!) and take it from there. Anyone have any
improvements and/or additions? Apologies for the line wrap. No idea how to
prevent that in Thunderbird. The filters are also at
http://pastebin.com/6T9M1W3F
Not sure but it may be possible that logging has changed between
Asterisk 1.4, 1.6, 1.8 and 10 so please mention the asterisk version with
your filters.
For Asterisk 1.8:
failregex = Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
No matching peer found
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Device does not match ACL
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Username/auth name mismatch
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Peer is not supposed to register
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' (from HOST)
NOTICE.* .*: Host HOST failed MD5 authentication for '.*'
(.*)
VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice'
(language '.*')
There are 2 lines that I have which are not in this list:
NOTICE.* .*: Registration from '.*' failed for 'HOST' - ACL error
(permit/deny)
NOTICE.* .*: Failed to authenticate user .*@HOST.*
How about those (no idea for which Asterisk version they are)?
Regards,
Patrick
Thanks Patrick. This is a great initiative. Let's all build the strongest
and most detailed filter possible. I actually looked at mine and now see
that it has weaknesses due Asterisk 1.8.8x giving different type of logs or
maybe FreePBX. Let's test, fix and append to the end of the filter.
Everyone is welcome to contribute.
So far we have:
*For Asterisk 1.8:*
failregex = Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No
matching peer found
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Device does not match ACL
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Username/auth name mismatch
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Peer
is not supposed to register
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' (from HOST)
NOTICE.* .*: Host HOST failed MD5 authentication for '.*' (.*)
VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice'
(language '.*') *#Outdated?*
#*Situation:* allowguest=yes and context=default in sip.con - *Tested
by **Diego Aguirre?*
NOTICE.* .*: Call from '.*' (HOST) to extension '.*' rejected
because extension not found in context 'default'
The following are what I found to be insecure but need escaping and fine
tuning to work with filter:
*Asterisk 1.8 + FreePBX:*
*Situation:* When target is coming in from unknown DID -
Needs character escaping
Executing [unknown@from-sip-external:1] NoOp(SIP/10.0.0.6-0001,
Received incoming SIP connection from unknown peer to unknown) in new
stack
*Situation:* Same as above except for an extension is called. Above was
just IP call. Extension 011x doesn't exist.
Executing [011566@from-sip-external:1] NoOp(SIP/10.0.0.6-0003,
Received incoming SIP connection from unknown peer to 011566) in new
stack
*Situation: *Same as above except for extension 101 does exist but system
still rejects calls due to no guest allowed?!
Executing [101@from-sip-external:1] NoOp(SIP/10.0.0.6-0005, Received
incoming SIP connection from unknown peer to 101) in new stack
*All of above have this following which can be used as a universal
filter: *Executing
[s@from-sip-external:8] Playback(SIP/10.0.0.6-0005, ss-noservice)
in new stack *
*
*
***Notice how this ss-noservice is difference from current the outdated
filter one:
*VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice' (language
'.*')*
-Bruce
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[asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet
Hi everyone, I see that there was a bug in version 1.8.5.x and people were advised to move to 1.8.7.1 but now I have 1.8.7.1 and experiencing the same problem. Here is the output: *chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729)* * * Now, I see an update to 1.8.8.1 is available. I am wondering if this issue is fixed in this version or not? Furthermore, has anyone tested 1.8.8.1 yet? Are there any other problems to that? It's frustrating as I see we should once again move back to 1.6x and forget about 1.8x all together. Any input is 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 1.8.7.1 forcing uLaw bug NOT fixed yet
So, what is really the effect of this and why is it hard to fix? Does this
bug disrupt processing the call? I see the log filled up with this error. I
do have a BUSY showing on forwarding to a number outside and that is what
concerns me. Not sure if caused by this bug. From reading CHANGES log, I
see that this has to do something with oversize packets in g729. Maybe it's
a setting issue?
Regards,
On Wed, Dec 28, 2011 at 3:19 PM, Danny Nicholas da...@debsinc.com wrote:
This might or might not help, but here is the offending code in 1.8.8
case AST_FRAME_VOICE:
if (!(frame-subclass.codec ast-nativeformats)) {
char s1[512], s2[512], s3[512];
ast_log(LOG_WARNING, Asked to transmit frame type
%s, while native formats is %s read/write = %s/%s\n,
ast_getformatname(frame-subclass.codec),
ast_getformatname_multiple(s1, sizeof(s1),
ast-nativeformats AST_FORMAT_AUDIO_MASK),
ast_getformatname_multiple(s2, sizeof(s2),
ast-readformat),
ast_getformatname_multiple(s3, sizeof(s3),
ast-writeformat));
and the comparable code in 10.0.0
case AST_FRAME_VOICE:
if (!(ast_format_cap_iscompatible(ast-nativeformats,
frame-subclass.format))) {
char s1[512];
ast_log(LOG_WARNING, Asked to transmit frame type
%s, while native formats is %s read/write = %s/%s\n,
ast_getformatname(frame-subclass.format),
ast_getformatname_multiple(s1, sizeof(s1),
ast-nativeformats),
ast_getformatname(ast-readformat),
ast_getformatname(ast-writeformat));
I personally avoided the 1.6 and 1.8 branches like the plague and don't
know
if this bug is corrected by the other fixes in 10.0.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, December 28, 2011 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed
yet
The issue is not fixed in 1.8.8.0 either.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Wednesday, December 28, 2011 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet
Hi everyone,
I see that there was a bug in version 1.8.5.x and people were advised to
move to 1.8.7.1 but now I have 1.8.7.1 and experiencing the same problem.
Here is the output:
chan_sip.c: Asked to transmit frame type ulaw, while native formats is
0x100 (g729) read/write = 0x100 (g729)/0x100 (g729)
Now, I see an update to 1.8.8.1 is available. I am wondering if this issue
is fixed in this version or not? Furthermore, has anyone tested 1.8.8.1
yet?
Are there any other problems to that? It's frustrating as I see we should
once again move back to 1.6x and forget about 1.8x all together.
Any input is appreciated.
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Re: [asterisk-users] 1.6 and 1.8
I have been running 1.8.7 with a few fixes back ported from the 1.8.8 release candidate for the last 2.5 months. The system processes around 4,000 calls per day over PRIs for 250 Polycom phones. Previously I was running 1.6.1.18 with a bunch of back ports for fixes and features. Overall it was stable but every few months I had an issue where a channel would get hung. When this happened core show channels would crash the console and I would eventually have to restart Asterisk. Ryan What od you mean by, been running 1.8.7 with a few fixes back ported from the 1.8.8 release candidate. So, this is a version 1.8.7 release that you are using or a 1.8.8 or is this a mix of both that you come up with? Can you please be specific with fixes? 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
Re: [asterisk-users] Interesting attack tonight fail2ban them
You mentioned the IP, 208.122.57.58, where did you get that from?
Following are the default for Asterisk 1.8 (It would be great to have
others input on this to strengthen this part of the filter):
failregex = Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No
matching peer found
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Device does not match ACL
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Username/auth name mismatch
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Peer
is not supposed to register
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' (from HOST)
NOTICE.* .*: Host HOST failed MD5 authentication for '.*' (.*)
VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice'
(language '.*')
Regards,
On Wed, Dec 28, 2011 at 11:50 PM, Michelle Dupuis mdup...@ocg.ca wrote:
I just realized there is no IP (host) in the message line, so no way for
fail2ban to catch it.
Other suggestions? Or will I have to code something into my dialplan
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[asterisk-users] A lot of 603 Declined Error form iCall - Are they going down or is it just bad service?
Hi everyone, Since three weeks ago, we have been getting A LOT of 603 Declined calls from iCall. I called a few times and their support is either non-responsive (they never call back) or can't fix the issue. I am wondering if everyone else is experiencing the same thing or is it because we recently upgraded from Asterisk 1.6x to Asterisk 1.8x and there is something that is causing this. This happens to their domestic and international routes. I would appreciate the input from those who use their services. 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
Re: [asterisk-users] No rtpmap codec info in 200 OK
I could be wrong but this sounds like a NAT issue rather SIP related packet issue. You are not receiving a response back is what I get a lot of times when my NAT is not setup properly. Call goes on for 10 or 20 second (I try the echo application and it hangs up before I get to talk) and then cuts off. -Bruce On Mon, Dec 19, 2011 at 7:41 PM, William Scott will...@magicwilly.infowrote: It seems quite unlikely that the presence of an 'a=rtpmap' line in the SDP for G.729 is going to cause a call to have any problems. Thanks for the reply. I'll expand on the scenario... This particular ATA does not send 'a=rtpmap' for any codec. When talking to a Asterisk PBX everything works fine. When talking to a VSP that sends an INVITE with User-Agent: Sippy the call is setup then drops after 32 seconds. Packet captures shows that no ACK is received after the ATA sends the 200 OK (missing rtpmap). After sending 200 OK about 6 times it then sends BYE and the call disconnects. Every other ATA I have sends rtpmap and works fine. The idea was to manipulate Asterisk into not sending rtpmap for the codec to confirm what happens. I'll now look for another solution. -- _ -- 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
Re: [asterisk-users] No rtpmap codec info in 200 OK
Can you register with Eyebeam to VSP and have it work? Make sure you are on the exact same network as the ATA when making this test. This should isolate the NAT issue. On Mon, Dec 19, 2011 at 9:27 PM, William Scott will...@magicwilly.infowrote: On 20 December 2011 12:51, Bruce B bruceb...@gmail.com wrote: I could be wrong but this sounds like a NAT issue rather SIP related packet issue. I looked at this to start with. Spent sometime comparing addresses and ports between successful and failure packets. Couldn't see any ports that weren't opened on the way out or the use of private ip addresses. I cleared the nat translation table between tests. This ATA works fine with Asterisk based VSPs. I'm just going to have to get more methodical. FYI, the ATA is a GW211 (mass produced OEM device, this one labelled Cormain) and the VSP is Pennytel here in Australia. -- _ -- 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
Re: [asterisk-users] FreePBX not updating configs on 1.8 RPM install
You needed to do asterisk -g or amportal start after your install. The configs didn't apply because Asterisk wasn't running so there was no connection to AMI. But when you updated module you Fpbx did an amportal restart or start automatically and hence it worked. Anyhow, but the FPBX rpm is broken and flawed from Digium. There are serious unsolved bugs on it with no useful response. On Fri, Dec 16, 2011 at 3:27 PM, Eric Germann egerm...@limanews.com wrote: Answering my own question, which is probably bad form. Updated the modules to current (from 2.7.0.0), applied config, now it works. Odd. EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Germann Sent: Friday, December 16, 2011 3:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FreePBX not updating configs on 1.8 RPM install Thanks. Checked. Both running as 'asterisk' EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, December 16, 2011 3:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FreePBX not updating configs on 1.8 RPM install Confirm your web server user is running as the same user as asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Germann Sent: Friday, December 16, 2011 3:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FreePBX not updating configs on 1.8 RPM install Brand new instance on Centos 5.7 Installed asterisk18 via yum from RPM distribution from Digium Installed FreePBX via yum from Digium distribution. Asterisk is up. FreePBX is up. However, the changes made in FreePBX aren't written out to the config files in /etc/asterisk nor does asterisk recognize any of the configs. Am I missing something? Been a little while since I installed them, but don't recall it being this difficult. Thoughts? EKG -- _ -- 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 -- _ -- 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] SIP MESSAGE outside calls - state of the art?
I think it only works with certain soft phones. I tried Aastra and it doesn't work. But EyeBeam soft phone receives messages. -Bruce On Mon, Dec 12, 2011 at 6:40 PM, Jay R. Worthington jayrworthing...@gmail.com wrote: Hiya, SIP Messaging is implemented in asterisk-10... The only documentation I can find talks about a patch and is pretty old:http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging Like anything on voip-info.org it's horrible outdated. I think there's a documentation for the message-routing in docs Regards Jay -- _ -- 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
Re: [asterisk-users] how to know RTP por of a SIP client in
Then you may use system() in dial-plan to run that shell command along with
what I suggested.
-Bruce
On Sun, Oct 23, 2011 at 5:22 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Yes, I need to know to get in in dialplan because I want to capture traffic
per call. I would like to launch $SHELL{tcpdump src port } in the
dialplan or something like this. And I want RTP traffic only of a certain
call.
Thank you!
===
Date: Fri, 21 Oct 2011 09:41:39 -0400
From: Bruce B bruceb...@gmail.com
Subject: Re: [asterisk-users] how to know RTP por of a SIP client in
the dialplan
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
CAJyE_uWLRXkhrWQ-6SvNW1ihN-nGA3HFwHt=pu-tfr6lybi...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Do you need to know to get it in dialplan? If I not, from shell (not
Asterisk CLI) I usually use:
netstata -a | grep asterisk
By default Asterisk settings it should be something between 10k-20k
-Bruce
On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Hi all,
How can I get the RTP port one SIP client is using for sending/receiving
RTP flow? Can I obtain it in from SIP_HEADER of something like that in
the
dialplan?
Thank you!
** **
Isabel
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Re: [asterisk-users] how to know RTP por of a SIP client in the dialplan
Do you need to know to get it in dialplan? If I not, from shell (not Asterisk CLI) I usually use: netstata -a | grep asterisk By default Asterisk settings it should be something between 10k-20k -Bruce On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL i...@tid.es wrote: Hi all, How can I get the RTP port one SIP client is using for sending/receiving RTP flow? Can I obtain it in from SIP_HEADER of something like that in the dialplan? Thank you! ** ** Isabel -- 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 -- _ -- 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
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
Paul, These trolls are the people who put your kid to school and put food on your table by giving valuable input and testing the open source software. Are you sure Digium endorses this stand of yours? Does everyone at Digium think the users who gives feedback that is not exactly what you like is a troll? WOW! I thought only rogue users try to censor this list but congratulations to Digium's own employees. Антон, Thanks. I will explore the option. -Bruce On Sun, Sep 25, 2011 at 12:05 PM, Paul Belanger pabelan...@digium.comwrote: On 11-09-25 01:54 AM, Антон Квашёнкин wrote: Just use cli aliases, provided by res_clialiases.so. 2011/9/25 Bruce Bbruceb...@gmail.com Please don't feed the trolls. Thanks. -- 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-**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
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
Paul, LOL...you are trying to change the subject. That's naive. You clearly know that I complained that there is no need for such drastic changes and long commands. The fact that it's written in CHANGES file or if there was a commit for it doesn't make it any better. Stop with the flawed reasoning. I am not going to complement your code or policies the whole time. Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. All I am saying is that - Come up with a naming convention and for the sake of everyone stick to it. How hard could that be? Even with new features you can still stick to certain principles if you plan it ahead. If you don't know how to do it, ask the community for input and people will help. -Bruce On Sun, Sep 25, 2011 at 1:22 PM, Paul Belanger pabelan...@digium.comwrote: On 11-09-25 01:01 PM, Bruce B wrote: Paul, These trolls are the people who put your kid to school and put food on your table by giving valuable input and testing the open source software. Are you sure Digium endorses this stand of yours? Does everyone at Digium think the users who gives feedback that is not exactly what you like is a troll? WOW! I thought only rogue users try to censor this list but congratulations to Digium's own employees. Антон, Thanks. I will explore the option. If you had bothered to search or even look at the CHANGES file, located in the source directory of asterisk, you would have seen the following: * Cleanup another bunch of CLI commands. Now all modules follow the same schema. (Done by lmadsen, junky and mvanbaak during the devcon 2008) Additionally, you could have taken the time to actually find the commit that made the change, since this is open source software everything is listed online [1]. Which was done by mvanbaak, an asterisk community member, not a Digium employee. [1] http://svnview.digium.com/svn/**asterisk?view=revision** revision=145121http://svnview.digium.com/svn/asterisk?view=revisionrevision=145121 -- 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-**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
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
You are very childish besides being very useless. Also, note that there are others that are bothered by the same changes that are uncalled for. I was as constructive as possible but you think starting a sentence with I am not trying to be rude... is rude. LOL. I have said that upfront so idiots like you don't take offence but you did and you read as, I am trying to be rude Well, suit yourself and keep sucking up Alex. On Sun, Sep 25, 2011 at 4:41 PM, Alex Balashov abalas...@evaristesys.comwrote: On 09/25/2011 02:23 PM, Bruce B wrote: Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. You won't get an audience if the way you go about it is dickish. You're being a dick, and you know you're being a dick. You're just unwilling to admit it or intellectually engage with that. If you were earnest and sincere about your desire to contribute constructive criticism and effectuate change, you wouldn't start the thread with a sarcastic subject line like Who is the 'creative' mind behind changing Asterisk commands at CLI? That has a mocking, derisive inflection, and you know it has a mocking, derisive inflection. If you expect to be taken seriously, you need to align your behaviour with your stated objective--unless that's not actually your objective, and in fact your objective is to be an inflammatory jerk. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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-**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
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
First of all, what the heck is this link you referenced: http://lists.digium.com/**pipermail/asterisk-users/2010-**April/247084.htmlhttp://lists.digium.com/pipermail/asterisk-users/2010-April/247084.html ?? Secondly, an Asterisk 1.6.2.18 that I am running right now plays nicely with help command. The 1.8 does do that. So, 1.6.2.18 has not been around for 3 years. Again, stop misleading and changing the subject. When you state 3 years ago that is absolutely false. In doesn't apply to any of the Asterisk versions till 1.8xx My post was very clear. Yes, it was sarcastic due to frustration but it was very clear and I wanted to say that there is no need to do core show help sip when you can simply do help sip. I still don't think your reply was called for. These trolls like I said help you live through with your attitude. If you were my employee and talked like this to anyone I would fire you right away. I am asking you nicely to please stop making this about yourself or Digium. Like I said, I like Asterisk. I love it. It works very good. Please listen to the community feedback without getting so defensive. No one gains anything from changes like this. I am sure Digium can afford one afternoon meeting to decide what the commands naming convention should be for the next 20 years. On Sun, Sep 25, 2011 at 4:05 PM, Paul Belanger pabelan...@digium.comwrote: On 11-09-25 02:23 PM, Bruce B wrote: Paul, LOL...you are trying to change the subject. That's naive. You clearly know that I complained that there is no need for such drastic changes and long commands. The fact that it's written in CHANGES file or if there was a commit for it doesn't make it any better. Stop with the flawed reasoning. I am not going to complement your code or policies the whole time. Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. All I am saying is that - Come up with a naming convention and for the sake of everyone stick to it. How hard could that be? Even with new features you can still stick to certain principles if you plan it ahead. If you don't know how to do it, ask the community for input and people will help. -Bruce You do realize this change happen almost 3 years go, aprox Nov. 2008. There was a discussion about it at Astricon, on -dev mailing list, plus a code review on reviewboard[1]. Implying it did not happen is incorrect. You might not have know about it because your first post from bruceb...@gmail.com seems to be Apr. 2010[2]. Community feedback was provided for the change, since it was driven by the community. If you don't like the change and want it reverted, simply load res_clialiases.so and edit cli_aliases.conf. Voicing your opinions is not a problem, however starting them with 'I don't mean to be rude but...' is not the best way to start them. If you want to help shape the future of Asterisk, I encourage you to join the discussion on the asterisk-dev mailing lists. Its open source software, everybody gets a say. It doesn't mean it will get done however. [1] https://reviewboard.asterisk.**org/r/32/https://reviewboard.asterisk.org/r/32/ [2] http://lists.digium.com/**pipermail/asterisk-users/2010-** April/247084.htmlhttp://lists.digium.com/pipermail/asterisk-users/2010-April/247084.html -- 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-**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
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
Thank you for a constructive reply. I am not a war monger and I appreciate a proper response. I will explore my options to that. My opinion may still be that such long commands are unnecessary but at least it seems there is a way to go around them for now and I am happy to hear that. On Sun, Sep 25, 2011 at 9:23 PM, Paul Belanger pabelan...@digium.comwrote: On 11-09-25 08:57 PM, Bruce B wrote: First of all, what the heck is this link you referenced: http://lists.digium.com/pipermail/asterisk-users/2010-** **April/247084.htmlhttp://lists.digium.com/**pipermail/asterisk-users/2010-**April/247084.html http://**lists.digium.com/pipermail/**asterisk-users/2010-April/** 247084.htmlhttp://lists.digium.com/pipermail/asterisk-users/2010-April/247084.html ?? Secondly, an Asterisk 1.6.2.18 that I am running right now plays nicely with help command. The 1.8 does do that. So, 1.6.2.18 has not been around for 3 years. Again, stop misleading and changing the subject. When you state 3 years ago that is absolutely false. In doesn't apply to any of the Asterisk versions till 1.8xx You seem to be missing the point or not reading my replies. The reason '*CLI help' still works on asterisk 1.6.2, is because of the changes made 3 years ago add res_clialiases.so. Without it, the command would actually not work. Here is a simple test you can do on your 1.6.2 / 1.8 asterisk box: *CLI module unload res_clialiases.so Unloaded res_clialiases.so *CLI help No such command 'help' (type 'core show help help' for other possible commands) As you can see, without res_clialiases.so the command does not work. So, if you are saying the '*CLI help' command does not work, then check your asterisk configuration first. My post was very clear. Yes, it was sarcastic due to frustration but it was very clear and I wanted to say that there is no need to do core show help sip when you can simply do help sip. I still don't think your reply was called for. These trolls like I said help you live through with your attitude. If you were my employee and talked like this to anyone I would fire you right away. I am asking you nicely to please stop making this about yourself or Digium. Like I said, I like Asterisk. I love it. It works very good. Please listen to the community feedback without getting so defensive. No one gains anything from changes like this. I am sure Digium can afford one afternoon meeting to decide what the commands naming convention should be for the next 20 years. I don't even know how to reply to this, so I won't. Thanks for all the fish. -- 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-**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
[asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
Hi everyone, I don't mean to be rude but honestly which genius comes up with changing the simple: help to core show help That's just an example. If it was only this or if this was only a two words loss then I would be fine. I think someone just loves to play around with the commands with each and every release. 1 word turned into 3 long words for a simple simple simple help command. Good job Digium. Cheers, -- _ -- 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] cli command show codecs
if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If your IVR announcement is not recorded in g729 and you see g729 on the channel when you call into IVR then it's transcoding as well. On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling ewiel...@nyigc.com wrote: Assuming SIP sip show channels will show you which codec is used for each call leg. However it does not track transcoding. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs asterisk -rx core show channels verbose does not provide transcoding details. Unless I have missed something. Sans On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas da...@debsinc.com wrote: Core show channels verbose is probably your best bet. I think the answer also depends on your * version. From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cli command show codecs Hi, Is there a CLI command which will tell me the codec used for active calls and if transcoding is happening ? Thx Sans -- _ -- 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
Re: [asterisk-users] cli command show codecs
sip show channels is the command you are looking for. On Wed, Aug 31, 2011 at 2:45 PM, RSCL Mumbai rscl.mum...@gmail.com wrote: asterisk -rx core show channels verbose does not provide transcoding details. Unless I have missed something. Sans On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas da...@debsinc.comwrote: Core show channels verbose is probably your best bet. I think the answer also depends on your * version. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *RSCL Mumbai *Sent:* Wednesday, August 31, 2011 10:44 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] cli command show codecs ** ** Hi, Is there a CLI command which will tell me the codec used for active calls and if transcoding is happening ? Thx Sans -- _ -- 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
[asterisk-users] Softhungup missing from Asterisk 1.6.20-1 - *without any notice*
Hello, Is softhangup still there? It's unknown command to Asterisk 1.6.20-1..there is no mention of this in CHANGES files. Also channel hangup request SIP/channel-name doesn't work for SIP. Is there any other command I am missing? 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
[asterisk-users] What sort of information does LIDB provide?
Hi everyone, In USA when doing a CNAM search, what sort of information is provided back? Does this information include carrier name? service address? service type (public or private phone)? etc...? Also, if you are not a CLEC do you have to purchase this service through a mediator CNAM look provider or is LIDB access open to everyone? 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
[asterisk-users] How is a ping test delay ms different from status in Asterisk sip show peers?
Hi everyone, Pinging a phone set I get 0.529 ms round trip delay. Running sip show peers in Asterisk CLI I see anywhere from 5 milli seconds to 280 ms. How are both of these different and why are they so different? Is the latter based on SIP packets return? I have a paging device that shows close to 280 ms which is not right but at ping it's 0.5 ms. Regards, -- _ -- 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] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
Can you please elaborate on how to apply the patch? Also, is the repository updated with the new code? Regards, On Tue, Aug 2, 2011 at 7:34 PM, Richard Mudgett rmudg...@digium.com wrote: Can you please point me to the patch that you just made? The patch is committed to v1.6.2 SVN branch. Patch for v1.6.2 only. r330490 | jrose | 2011-08-01 16:08:10 -0500 (Mon, 01 Aug 2011) | 12 lines Asterisk 18103 - Fix reload crash caused by destroying default parking lot Default parking lot was being destroyed in reload and was not being rebuilt properly. This patch keeps features.c reload from destroying the default parking lot in 1.6.2. Bug was caused by a hasty backport which didn't test reload enough times to catch the problem. (closes issue ASTERISK-18103) Reported by: 808blogger Review: https://reviewboard.asterisk.org/r/1337/ Also -r330505 to fix a ref leak with the above patch. Richard -- _ -- 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
Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
I am happy it's being taken care of. Would the patch fix systems that used the Repo to install Asterisk 1.6.2.19? That is where we all have problems. Or maybe a new version of Asterisk which yum update would do the job? On Mon, Aug 1, 2011 at 11:27 PM, Jonathan Rose jr...@digium.com wrote: We worked on this bug today and are expecting to release packages with the fix soon, possibly tomorrow (Aug 2). The issue arose from a change in features reload which was back-ported to 1.6.2 and was committed without enough testing to observe the intermittent crash behavior. Thanks for your patience, Jonathan R. Rose - Original Message - From: Vahan Yerkanian va...@arminco.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 31, 2011 3:03:07 AM Subject: Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs? On 7/30/11 7:39 AM, Bruce B wrote: I think this should be a quick fix since it's rendering the latest stable version useless and making the impression that it was released just to break things and force people onto 1.8x. Just a thought...no blame game. But really something like this should be tackled quickly. No point to break things so badly on the last stable version. Regards, Confirming this issue on 10+ 1.6.2.19 Asterisk servers. This problem makes it difficult to do edits on sip.conf on production systems, as there is ~25% chance that you'll crash the server and cut the established calls. The problem does not exist in 1.6.2.18... I think this problem should be fixed or the 1.6.2.19 should be removed from the digium repo. Regards, Vahan -- _ -- 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
Re: [asterisk-users] use ILBC installed from asterisk yum repositories
I would be very interested in iLBC. I even posted regarding this to this mailing list and the thread died after no one was able to confirm it works. I think there are others who would really like to see H.323 working from the repo as well (I think that is not working as well). Regards, On Tue, Aug 2, 2011 at 12:46 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 08/02/2011 11:42 AM, Bob Pierce wrote: I would like to try the ILBC codec on one of our systems. The system is currently running Asterisk 1.8.5.0 installed from the Asterisk provided repositories for Centos 5. Is there a process for installing the ILBC codec under this environment, or will I have to un-install the RPMs and build Asterisk from source? There is no codec_ilbc RPM available from the Digium repositories at this point; there could be one in the future, but given that this is the first time I've seen a request for it, it seems unlikely to be worth the effort. You can use the SRPM for Asterisk to rebuild the RPM after importing the iLBC source into the build tree; at least I think that would work. -- 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 -- _ -- 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] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
Hi Jonathan, Any clue with 1.6.2.19.*1 *might be released? Regards, On Mon, Aug 1, 2011 at 11:27 PM, Jonathan Rose jr...@digium.com wrote: We worked on this bug today and are expecting to release packages with the fix soon, possibly tomorrow (Aug 2). The issue arose from a change in features reload which was back-ported to 1.6.2 and was committed without enough testing to observe the intermittent crash behavior. Thanks for your patience, Jonathan R. Rose - Original Message - From: Vahan Yerkanian va...@arminco.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 31, 2011 3:03:07 AM Subject: Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs? On 7/30/11 7:39 AM, Bruce B wrote: I think this should be a quick fix since it's rendering the latest stable version useless and making the impression that it was released just to break things and force people onto 1.8x. Just a thought...no blame game. But really something like this should be tackled quickly. No point to break things so badly on the last stable version. Regards, Confirming this issue on 10+ 1.6.2.19 Asterisk servers. This problem makes it difficult to do edits on sip.conf on production systems, as there is ~25% chance that you'll crash the server and cut the established calls. The problem does not exist in 1.6.2.18... I think this problem should be fixed or the 1.6.2.19 should be removed from the digium repo. Regards, Vahan -- _ -- 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
Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
Can you please point me to the patch that you just made? Thanks On Mon, Aug 1, 2011 at 11:27 PM, Jonathan Rose jr...@digium.com wrote: We worked on this bug today and are expecting to release packages with the fix soon, possibly tomorrow (Aug 2). The issue arose from a change in features reload which was back-ported to 1.6.2 and was committed without enough testing to observe the intermittent crash behavior. Thanks for your patience, Jonathan R. Rose - Original Message - From: Vahan Yerkanian va...@arminco.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 31, 2011 3:03:07 AM Subject: Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs? On 7/30/11 7:39 AM, Bruce B wrote: I think this should be a quick fix since it's rendering the latest stable version useless and making the impression that it was released just to break things and force people onto 1.8x. Just a thought...no blame game. But really something like this should be tackled quickly. No point to break things so badly on the last stable version. Regards, Confirming this issue on 10+ 1.6.2.19 Asterisk servers. This problem makes it difficult to do edits on sip.conf on production systems, as there is ~25% chance that you'll crash the server and cut the established calls. The problem does not exist in 1.6.2.18... I think this problem should be fixed or the 1.6.2.19 should be removed from the digium repo. Regards, Vahan -- _ -- 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
Re: [asterisk-users] use ILBC installed from asterisk yum repositories
There is much more to installing and configuring OOH323 as it's not easy breezy install. I think a professional developer help would be more appropriate than users patching. Just my thought.plus it adds a great deal of functionality to Asterisk to allow for all add-ons to be install via RPMS or at least the ones related to codec and protocols. On Tue, Aug 2, 2011 at 6:33 PM, Bob Pierce westman...@gmail.com wrote: On Tue, Aug 2, 2011 at 4:39 PM, Lefteris Zafiris zaf@gmail.com wrote: You can write a short makefile for just codec_ilbc module, build it and install it on your running asterisk system. You will have to install the asterisk18-devel package and get the asterisk source code either from a tar or from the srpm. If you are familiar with the basics of writing makefiles its pretty trivial to write one that builds codec_ilbc, I have done this in numerous systems that use the digium rpms and it works flawlessly. This method can also be used to build other modules that are missing from the digium rpms. Thanks for the pointer. I think I'll give this method a try. I'll see if I can figure out how to write the makefile. -- _ -- 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
Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
Thanks for feedback. Yeah, tell me about it. Your description is very accurate of the situation. I can't believe it's in the repo without any tests done; even the simplest reload. I don't mean to be a whiner but honestly the repo is a joke with such an obvious flaw for so long now On Sun, Jul 31, 2011 at 4:03 AM, Vahan Yerkanian va...@arminco.com wrote: On 7/30/11 7:39 AM, Bruce B wrote: I think this should be a quick fix since it's rendering the latest stable version useless and making the impression that it was released just to break things and force people onto 1.8x. Just a thought...no blame game. But really something like this should be tackled quickly. No point to break things so badly on the last stable version. Regards, Confirming this issue on 10+ 1.6.2.19 Asterisk servers. This problem makes it difficult to do edits on sip.conf on production systems, as there is ~25% chance that you'll crash the server and cut the established calls. The problem does not exist in 1.6.2.18... I think this problem should be fixed or the 1.6.2.19 should be removed from the digium repo. Regards, Vahan -- __**__**_ -- 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
[asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
Hi everyone, Asterisk 1.6.2.19 has a bug per: https://issues.asterisk.org/jira/browse/ASTERISK-18103 What is the general time to fix this? I think a similar thing is also noted in 1.8x install. Is it not going to be taken care of because it's 1.6x ? 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
Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
I think this should be a quick fix since it's rendering the latest stable version useless and making the impression that it was released just to break things and force people onto 1.8x. Just a thought...no blame game. But really something like this should be tackled quickly. No point to break things so badly on the last stable version. Regards, On Fri, Jul 29, 2011 at 6:23 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 07/29/2011 06:20 PM, Paul Belanger wrote: On 11-07-29 06:12 PM, Bruce B wrote: Hi everyone, Asterisk 1.6.2.19 has a bug per: https://issues.asterisk.org/**jira/browse/ASTERISK-18103https://issues.asterisk.org/jira/browse/ASTERISK-18103 What is the general time to fix this? I think a similar thing is also noted in 1.8x install. Is it not going to be taken care of because it's 1.6x ? 1.6.2.19 was to be the last release of the 1.6.2 branch, so I'm not sure if another build is expected. However the issue does reference 1.6.2.19.1 so it is possible. However, you can see what changed between 1.6.2.18 and 1.6.2.19 in an attempted to narrow down the bug. If it was a regression from 1.6.2.18 to 1.6.2.19, then it will be fixed. -- 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 -- _ -- 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] Securing Asterisk
Hmmm, if alwaysauthreject is already breaking RFC rules then why not break another rule for the greater good? It would only add another layer of security. Maybe: *alwaysregreject=yes* * * *To drop SIP packets for both unauthorized registers and anonymous calls. Keep it off by default and then allow users to turn it on if they want to. To be fair to OP, using Asterisk with open ports to the world is a legit use of Asterisk even if most of us don't employ it that way or use it solely with closed networks (VPN, etc...). There are many people who would benefit from a security feature that would simply ignore unauthorized registers and anonymous calls. OP is suggesting an improvement to Asterisk; maybe people should weigh options and see if it's time to act more on the security side or not. There is no question that if a hacker knows there is a SIP server then they will keep the IP on the list for later use or share it with colleagues even if it seems secure right now. A DDoS is always a possibility and that you can't save yourself from at all. Right now the situation is more like this: *Knock Knock:* *Owner: *Whose there? *Thief:* This is Mr. X from China, and I am here to steal your TV. *Owner: *Hi, I am James Smith, 45, 190lbs and I have a nice laptop as well but I am home now and I can't let you in. *Thief (laughing):* No problem, I will come back at midnight when you are sleeping :-) - Bruce On Wed, Jul 27, 2011 at 2:20 PM, Matthew J. Roth mr...@imminc.com wrote: Kevin P. Fleming wrote: 'alwaysauthreject' in not imcompliant with any RFCs; the RFCs define response codes that *can* be used to indicate (for example) that the Request URI does not represent a target known to the receiver (404 Not Found), but does not mandate that the server respond with that code in that situation. Kevin, Thanks for the correction and I apologize if I'm propagating a misconception. Am I misunderstanding this Asterisk Security Advisory? http://lists.digium.com/pipermail/asterisk-announce/2009-April/000177.html In 2006, the Asterisk maintainers made it more difficult to scan for valid SIP usernames by implementing an option called alwaysauthreject... ...What we have done is to carefully emulate exactly the same responses throughout possible dialogs, which should prevent attackers from gleaning this information. All invalid users, if this option is turned on, will receive the same response throughout the dialog, as if a username was valid, but the password was incorrect. It is important to note several things. First, this vulnerability is derived directly from the SIP specification, and it is a technical violation of RFC 3261 (and subsequent RFCs, as of this date), for us to return these responses... I am asking out of genuine curiosity, because I trust your assessment more than my interpretation of the advisory. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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
Re: [asterisk-users] Securing Asterisk
I would have to err on the side of CDR to say that the only difference in analogy you provided (SSH vs Asterisk) is that people lose much more in VoIP than they ever did in SSH hacking. So, if this is an exceptional case bending a rule or two of RFC in favor of security won't harm specially if it's provided as an option. After-all, RFC does stand for Referral For Comment as in always open to be improved. Secondly, there is no trade off with the responses as local and private IP networks are well know from the public range so the option for such a security measure can be tuned to be smart to that end. The only thing I like about MS OSs is that it's secure out of box and that is really what a Linux OS should be as well but it's not and so it's not solely Digium's issue and I see your point giving the analogy. I think it's a good idea if such a security option is provided by default in Asterisk knowing it can save a lot of headache. If budget is an issue maybe make it a bounty and watch support pouring in... - Bruce On Tue, Jul 26, 2011 at 2:14 PM, Alex Balashov abalas...@evaristesys.comwrote: On 07/26/2011 02:09 PM, CDR wrote: Only way to cope with hackers would be that Digium comes to its senses and accepts to disable any response to a REGISTER whose username is unknown. I cannot think of a good reason why Digium finds this proposal unacceptable, given the onslaught of hacking that we are seeing in the industry. It may take a single line of code and it would save millions of $$$. Not only because the hackers will never get in, but because we would save a huge CPU impact responding to hundreds of REGISTER attempts per minute. It is a NO brainer. Can please the Powers that Be reconsider and add this option to sip.conf? Please? No, because that's absolutely ridiculous. The proper, RFC-compliant behaviour is to return an authentication failure in response to invalid credentials. This mechanism is relied upon for legitimate functionality, such as letting the UAs of intended users know that they are sending incorrect credentials. As was pointed out before, Asterisk is a mostly application-level construct. Applications usually have some rudimentary means of self-defense such as ACLs, but applications are often conceptually distinct from the most appropriate means of securing them. That's what firewalls, SBCs, intrusion detection systems, etc. are for. Your position is equivalent to saying that stock SSH should not return authentication errors for invalid passwords. The proper solution to dictionary attacks is to firewall the SSH service, use RSA keys, VPNs, etc., not to tell the maintainers of the OpenSSH project to come to its senses. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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-**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
Re: [asterisk-users] Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined
Not really. It's only good after DECLINED is sent. On Sat, Jul 23, 2011 at 2:08 AM, Mitesh Thakkar mail.mthak...@gmail.comwrote: I think fail2ban can help in this issue. Regards, Mitesh Thakkar +91 94279 07952 Yahoo: miteshthakkar...@yahoo.co.in GTalk: mail.mthak...@gmail.com On Sat, Jul 23, 2011 at 10:04 AM, Bruce B bruceb...@gmail.com wrote: Robert thanks for weighing in. So, you are saying that FreeSwitch on it's own can tackle issues like this without the need of OpenSIPs? Can you elaborate please? Thanks On Sat, Jul 23, 2011 at 12:17 AM, Robert-iPhone rhuddles...@gmail.com wrote: I like to put mine on 3389 hahaha just kidding. Personally I'm starting to convert to FreeSwitch - oops I had to say it. Security can be difficult and there are some good SBCs out there - just begs investment in technology - OH and bright staff Sent from my iPhone On Jul 23, 2011, at 12:09 AM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 22 Jul 2011, Bruce B wrote: 1- So, you are saying that either of OpenSER/Kamailio/OpenSIPS actually give me the full capability to the SIP stack to do the sort of thing I was asking for? And this can run on the same server as Asterisk is running? Configure OpenSIPS to listen to 5060 and Asterisk to listen to 5061. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- _ -- 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
[asterisk-users] Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined
Hello, I am wondering if there is a way to drop SIP packets for generic transactions? For example, only SIP PEERs are allowed to call in and receive ACK or Declined rather that those inviting a call who are not PEERs at all. Currently my Asterisk setup sends, *SIP/2.0 603 Declined *to any stranger invites because my dialplan includes Hangup(). Is there any way I can not send a 603 declined so to mislead the probe runner? 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
Re: [asterisk-users] Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined
Thanks for the input. I am really surprised. But yes, I want exactly what firewall does, DROP packet instead of REJECTING it. So, you are saying that one has to tamper the SIP stack to add the option to not respond to un-trusted sources? I really thought Asterisk might have this built in as a feature. I can't even do a dialplan search for a registered PEER because even if I find the IP to not be a trusted I still need to Hangup() on the invite which in turn send 603 Declined. There isn't really any work-around to this? Thanks again On Fri, Jul 22, 2011 at 7:39 PM, Alex Balashov abalas...@evaristesys.comwrote: On 07/22/2011 07:32 PM, Bruce B wrote: Hello, I am wondering if there is a way to drop SIP packets for generic transactions? For example, only SIP PEERs are allowed to call in and receive ACK or Declined rather that those inviting a call who are not PEERs at all. Currently my Asterisk setup sends, *SIP/2.0 603 Declined *to any stranger invites because my dialplan includes Hangup(). Is there any way I can not send a 603 declined so to mislead the probe runner? There is really no way to accomplish that except with a firewall. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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-**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
Re: [asterisk-users] Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined
Robert thanks for weighing in. So, you are saying that FreeSwitch on it's own can tackle issues like this without the need of OpenSIPs? Can you elaborate please? Thanks On Sat, Jul 23, 2011 at 12:17 AM, Robert-iPhone rhuddles...@gmail.comwrote: I like to put mine on 3389 hahaha just kidding. Personally I'm starting to convert to FreeSwitch - oops I had to say it. Security can be difficult and there are some good SBCs out there - just begs investment in technology - OH and bright staff Sent from my iPhone On Jul 23, 2011, at 12:09 AM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 22 Jul 2011, Bruce B wrote: 1- So, you are saying that either of OpenSER/Kamailio/OpenSIPS actually give me the full capability to the SIP stack to do the sort of thing I was asking for? And this can run on the same server as Asterisk is running? Configure OpenSIPS to listen to 5060 and Asterisk to listen to 5061. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867PST Newline Fax: +1-760-731-3000 -- _ -- 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
Re: [asterisk-users] Seg Faults with 1.6.2.19
I can confirm as well that there is an issue with Asterisk crashing. Asterisk 1.6.2.19 was installed using Digium repository. Probably some module was enabled in the repository install that is causing this. On Mon, Jul 18, 2011 at 12:13 PM, Lee Archer lee.arc...@thebigword.comwrote: Hi Kevin, the ticket below was closed as it doesn't happen with 1.8. It can't be related to my ODBC connections if others are having it. Should a new ticket be opened? Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: 18 July 2011 15:10 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19 On 07/18/2011 08:07 AM, Steve Davies wrote: On 18 July 2011 14:05, Lee Archerlee.arc...@thebigword.com wrote: Seems to be an already reported problem but since no more fixes for 1.6 it's back to 1.6.2.18.2 for me. https://issues.asterisk.org/jira/browse/ASTERISK-18103 Regards Lee If it is a regression introduced in 1.6.2.19, then it should still be fixed. At least I believe that's the rules. That should be the case, yes. -- 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-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
Re: [asterisk-users] Problem on Dialling-out
Yes, that is it. And you were inviting the provider to contact you back at your private subnet of 172.16.x.x: *From: Cordia sip:Unknown@172.16.9.15;tag=**as2267fdcc* * * So, hence their responces never made it back to you and that's why you are re-transmitting 6 times to get attention. * * - Bruce On Wed, Jul 13, 2011 at 2:49 AM, Malvin Rito mr...@mail.altcladding.com.phwrote: ** Bruce, Thanks. I already figured out the problem. It seems that a firewall issue. Regards, Malvin On 7/13/2011 12:30 PM, Bruce B wrote: Your trunk shows busy: * -- Called CordiaVoIP/639285010430 -- SIP/CordiaVoIP-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0)* Try this in the CLI (asterisk -r): *core set verbose 0* *sip set debug peer CordiaVoIP* And then make a call and read why the SIP trunk is failing. -Bruce On Wed, Jul 13, 2011 at 12:23 AM, Malvin Rito mr...@mail.altcladding.com.ph wrote: Hi List, I have a Asterisk + FreePbx Server setup with around 10 SIP extensions and 1 VoIP trunk (CordiaVoIP), when we dial-out to any number call is being dropped with the following message on asterisk log: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called CordiaVoIP/639285010430 -- SIP/CordiaVoIP-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/1001-0014, Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0) in new stack -- Executing [s@macro-dialout-trunk:21] Goto(SIP/1001-0014, s-CONGESTION,1) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set(SIP/1001-0014, RC=0) in new stack -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(SIP/1001-0014, 0,1) in new stack -- Goto (macro-dialout-trunk,0,1) -- Executing [0@macro-dialout-trunk:1] Goto(SIP/1001-0014, continue,1) in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] GotoIf(SIP/1001-0014, 1?noreport) in new stack -- Goto (macro-dialout-trunk,continue,3) -- Executing [continue@macro-dialout-trunk:3] NoOp(SIP/1001-0014, TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 0 - failing through to other trunks) in new stack -- Executing [continue@macro-dialout-trunk:4] Set(SIP/1001-0014, CALLERID(number)=1001) in new stack -- Executing [639285010430@from-internal:8] Macro(SIP/1001-0014, outisbusy,) in new stack -- Executing [s@macro-outisbusy:1] Progress(SIP/1001-0014, ) in new stack -- Executing [s@macro-outisbusy:2] Playback(SIP/1001-0014, all-circuits-busy-now,noanswer) in new stack -- SIP/1001-0014 Playing 'all-circuits-busy-now.gsm' (language 'en') -- Executing [s@macro-outisbusy:3] Playback(SIP/1001-0014, pls-try-call-later,noanswer) in new stack -- SIP/1001-0014 Playing 'pls-try-call-later.gsm' (language 'en') -- Executing [s@macro-outisbusy:4] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/1001-0014' in macro 'outisbusy' == Spawn extension (from-internal, 639285010430, 8) exited non-zero on 'SIP/1001-0014' -- Executing [h@from-internal:1] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-0014' localhost*CLI Can someone assist me please. Thanks in advance. Regards, Malvin -- _ -- 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
Re: [asterisk-users] Problem on Dialling-out
Your trunk shows busy: * -- Called CordiaVoIP/639285010430 -- SIP/CordiaVoIP-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0)* Try this in the CLI (asterisk -r): *core set verbose 0* *sip set debug peer CordiaVoIP* And then make a call and read why the SIP trunk is failing. -Bruce On Wed, Jul 13, 2011 at 12:23 AM, Malvin Rito mr...@mail.altcladding.com.ph wrote: Hi List, I have a Asterisk + FreePbx Server setup with around 10 SIP extensions and 1 VoIP trunk (CordiaVoIP), when we dial-out to any number call is being dropped with the following message on asterisk log: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called CordiaVoIP/639285010430 -- SIP/CordiaVoIP-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/1001-0014, Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0) in new stack -- Executing [s@macro-dialout-trunk:21] Goto(SIP/1001-0014, s-CONGESTION,1) in new stack -- Goto (macro-dialout-trunk,s-**CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-**trunk:1] Set(SIP/1001-0014, RC=0) in new stack -- Executing [s-CONGESTION@macro-dialout-**trunk:2] Goto(SIP/1001-0014, 0,1) in new stack -- Goto (macro-dialout-trunk,0,1) -- Executing [0@macro-dialout-trunk:1] Goto(SIP/1001-0014, continue,1) in new stack -- Goto (macro-dialout-trunk,continue,**1) -- Executing [continue@macro-dialout-trunk:**1] GotoIf(SIP/1001-0014, 1?noreport) in new stack -- Goto (macro-dialout-trunk,continue,**3) -- Executing [continue@macro-dialout-trunk:**3] NoOp(SIP/1001-0014, TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 0 - failing through to other trunks) in new stack -- Executing [continue@macro-dialout-trunk:**4] Set(SIP/1001-0014, CALLERID(number)=1001) in new stack -- Executing [639285010430@from-internal:8] Macro(SIP/1001-0014, outisbusy,) in new stack -- Executing [s@macro-outisbusy:1] Progress(SIP/1001-0014, ) in new stack -- Executing [s@macro-outisbusy:2] Playback(SIP/1001-0014, all-circuits-busy-now,**noanswer) in new stack -- SIP/1001-0014 Playing 'all-circuits-busy-now.gsm' (language 'en') -- Executing [s@macro-outisbusy:3] Playback(SIP/1001-0014, pls-try-call-later,noanswer) in new stack -- SIP/1001-0014 Playing 'pls-try-call-later.gsm' (language 'en') -- Executing [s@macro-outisbusy:4] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/1001-0014' in macro 'outisbusy' == Spawn extension (from-internal, 639285010430, 8) exited non-zero on 'SIP/1001-0014' -- Executing [h@from-internal:1] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-0014' localhost*CLI Can someone assist me please. Thanks in advance. Regards, Malvin -- __**__**_ -- 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:
[asterisk-users] No pattern 407 from SIP provider iCall
Hi everyone, Occasionally (with no set pattern), I get *SIP/2.0 407 Proxy Authentication Required *from iCall when trying to termiate to their international gateways. I have tried direct IP termination as well as SIP register but both just fail with above message whenever they want. Specially in register mode where the user is registered and both userid and password are good and they have been good yesterday, today they fail and the next day they work. What could be the reason? 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
[asterisk-users] Installing Asterisk from repository works great without the need to install Dahdi on Host Node of Proxmox - But trying to install from source fails. Why?
Hi everyone, I just lunched a CentOS VM in Proxmox and used the Digium repository to install Asterisk using yum install asterisk16...and it works great. Runs and it seems to have installed ztdummy as well without the need to touch the host node. But when I try to compile Dahdi from source on the same VM to install Asterisk from source I get this: #@root/usr/src/dahdi/: *make all* make -C linux all make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/firmware' *You do not appear to have the sources for the 2.6.32-4-pve kernel installed.* make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux' make: *** [all] Error 2 It seems that everyone is suggesting to install Dahdi on Host Node and then do modprobe ztdummy to get Dahdi running in VPS. Well, what is different between source install and repository install which doesn't need me to touch Host Node at all? I would rather not touch the Host Node at all and get a setup running just like Digium repository does. Any feedback is much 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
[asterisk-users] Clarification of the terms shown on CLI
Hi everyone, When doing a sip show settings on Asterisk 1.6.2.18, I see the following: Match Auth Username:No Allow unknown access: Yes Allow subscriptions:Yes Allow overlap dialing: Yes Allow promsic. redir: No Enable call counters: No What do each of above signify? 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
[asterisk-users] What is wrong in m
Hi everyone,
What is wrong in below asterisk application? The output should be content of
field booth_status from table booths:
[extension-status]
exten = _X.,1,MYSQL(Connect connid 127.0.0.1 root password my-extensions)
exten = _X.,n,MYSQL(Query allow_call ${connid} SELECT extension_status FROM
mytable WHERE extension=${CALLERID(num)} ORDER BY id DESC LIMIT 1)
exten = _X.,n,NoOp(allow_call is: ${allow_call})
But I get:
*allow_call is: 4 *while it should actually be ACTIVE or INACTIVE
I want to read the LAST record found under column extension in mytable
and do a NoOp for it's contents. But instead I am getting 4 which I think
refers to the connection ID?!
***There is only one record in my table right now.
Thanks,
--
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[asterisk-users] iptables for Asterisk - Any good guides out there?
Hi everyone, I want to issue the command: iptables -F and then rebuild everything from the beginning with a very limited scope and then without locking myself block all other traffic. Can you suggest what I should put in the shell that would get me this: Allow traffic from subnet 172.16.0.0/24 (my VPN tunnels) - All traffic including those of Asterisk and HTTP - I trust this network Allow traffic from subnet 192.168.1.0/24(other side of VPN network) - All traffic including those of Asterisk and HTTP - I trust this network Allow traffic from single IP of DID provider - 5060 TCP/UDP and 1-10200 UDP Allow VPN access on port 1194 UDP --- I have that figured out to be (*iptables -A INPUT -p udp -m udp --dport 1194 -j ACCEPT*) works for this. *BLOCK all other traffic - Important most of all* Please note that from the subnets I want to allow every single port possible and all traffic. I specially have problems with getting a whole subnet be able to access everything. 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
Re: [asterisk-users] iptables for Asterisk - Any good guides out there?
Thanks Hans. So basically run the following commands: iptables -P INPUT DROP iptables -P OUTPUT ACCEPT iptables -P FORWARD ACCEPT service iptables save iptables -F Is that all right so far? I am not sure on these: iptables -A INPUT -i $EXTERNAL_DEV -j LOG --log-prefix EXT; INC iptables -A OUTPUT -o $EXTERNAL_DEV -j LOG --log-prefix EXT; OUT iptables -A FORWARD -i $EXTERNAL_DEV -j LOG --log-prefix EXT; FWD And yes, of course I will need DHCP and all other necessary services to run the server. I am wondering why iptables is so complex. Is there a standard template that I can use to replace /etc/sysconfig/iptables with it and let it accept all traffic from one subnet on my tun0 which is my VPN and block all other traffic? Thanks again On Sat, May 14, 2011 at 8:14 PM, Hans Witvliet h...@a-domani.nl wrote: On Sat, 2011-05-14 at 19:51 -0400, Bruce B wrote: Hi everyone, I want to issue the command: iptables -F and then rebuild everything from the beginning with a very limited scope and then without locking myself block all other traffic. Can you suggest what I should put in the shell that would get me this: Allow traffic from subnet 172.16.0.0/24 (my VPN tunnels) - All traffic including those of Asterisk and HTTP - I trust this network Allow traffic from subnet 192.168.1.0/24(other side of VPN network) - All traffic including those of Asterisk and HTTP - I trust this network Allow traffic from single IP of DID provider - 5060 TCP/UDP and 1-10200 UDP Allow VPN access on port 1194 UDP --- I have that figured out to be (iptables -A INPUT -p udp -m udp --dport 1194 -j ACCEPT) works for this. BLOCK all other traffic - Important most of all Please note that from the subnets I want to allow every single port possible and all traffic. I specially have problems with getting a whole subnet be able to access everything. Thanks It's a bit more complicated Firstly you have to set the default rules FIRST $IPT -P INPUT DROP $IPT -P OUTPUT ACCEPT $IPT -P FORWARD ACCEPT And then do the flusing, not the otherway round After that you can add rules to accept trafic after the last rules, it is handy to put: $iptables -A INPUT -i $EXTERNAL_DEV -j LOG --log-prefix EXT; INC iptables -A OUTPUT -o $EXTERNAL_DEV -j LOG --log-prefix EXT; OUT iptables -A FORWARD -i $EXTERNAL_DEV -j LOG --log-prefix EXT; FWD So can can see in the syslog what you are missing ;-) I'll guess, you would also like to accepts ntp,dhcp, domain-dns from your isp-provider. Perhaps also http, https, pop, pops, imap, imaps. And probably some more, depending on your need So'll see them soon enough in your logfiles hw -- _ -- 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
Re: [asterisk-users] iptables for Asterisk - Any good guides out there?
Thanks Jeremy. But unfortunately no time to go over all this in detail. Maybe in future. Also because as I repeatedly said I have OpenVPN setup so I trust the VPN network there is no need for all this complication. Simply allowing all traffic out and only allowing VPN traffic in from tun0 would do for me. Thanks On Sat, May 14, 2011 at 9:46 PM, Jeremy Kister asterisk...@jeremykister.com wrote: On 5/14/2011 9:45 PM, Jeremy Kister wrote: http://jeremy.kister.net/code/asterisk/iptables.init oops, that's: http://jeremy.kister.net/code/iptables/iptables.init -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
Re: [asterisk-users] Occasional call from asterisk
Thanks for the input. Long ago the CDR showed asterisk as the CLID but it
doesn't anymore so I am puzzled now how to even stop taking calls because my
CLID is now blank and I can't refuse any call with no CLID.
*WARNING[11002] chan_dahdi.c: CallerID returned with error on channel
'DAHDI/2-1'*
Here are some out of place messages I am getting in my logs but nothing out
of norm around the time I get Ghost calls though:
*WARNING[11002] chan_dahdi.c: CallerID returned with error on channel
'DAHDI/2-1'*
*NOTICE[12524] chan_dahdi.c: Got event 17 (Polarity Reversal)...*
*
*
*
DEBUG[12524] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 4,
state 6
DEBUG[12524] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 4,
state 6
*
Can someone shed light on these options as to what exactly they do:
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
Hopefully some Asterisk guru can tell us more about what might be happening
as I see this as a situation that can be avoided or at least there should be
a workaround for this.
Regards,
On Mon, May 9, 2011 at 9:50 AM, Brian Henning bhenn...@pineinst.com wrote:
Hello Bruce,
I did not find a solution, only advice to lead me to think “huh, well
that’s annoying but we can deal with it.” I understand from my users,
though, that it’s *not* always the case that it’s a phantom call—sometimes
there really is someone calling.
Note that I haven’t tried what I’m about to suggest, but you might try
examining the CALLERID data before dialing the SIP extensions and, if it is
empty or contains “asterisk,” reset it to something like “not available.”
Cheers,
~Brian
*From:* Bruce B [mailto:bruceb...@gmail.com]
*Sent:* Friday, May 06, 2011 10:55 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Cc:* bhenn...@pineinst.com
*Subject:* Re: [asterisk-users] Occasional call from asterisk
Hi Brian,
Did you find a solution to your problem? or at least got a working
dial-plan for it? I have the same problem again as well and want to know
what to do with the dial-plan to off-set the effect at least since Telco
says it's not their issue.
Regards,
Bruce
On Thu, Apr 7, 2011 at 5:53 PM, Brian Henning bhenn...@pineinst.com
wrote:
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight?
Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,
Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup
The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings. Works great most of the time.
I have a hunch that maybe this happens if the inbound caller hangs up while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there. (The
preceding line of the log is a call from almost an hour earlier). In that
case though I'd expect to see 1-CANCEL in the log instead. Perhaps if
the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to
prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Thanks in advance for any and all advice!
~Brian
--
Brian Henning, Software Engineer
/\Pine Research Instrumentation
//\\ 5908 Triangle Drive
///\\\ Raleigh, NC 27617
USA
||
||phone: 919.782.8320
fax: 919.782.8323
email: bhenn...@pineinst.com
--
--
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Re: [asterisk-users] Occasional call from asterisk
Hi Brian,
Did you find a solution to your problem? or at least got a working dial-plan
for it? I have the same problem again as well and want to know what to do
with the dial-plan to off-set the effect at least since Telco says it's not
their issue.
Regards,
Bruce
On Thu, Apr 7, 2011 at 5:53 PM, Brian Henning bhenn...@pineinst.com wrote:
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight?
Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,
Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup
The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings. Works great most of the time.
I have a hunch that maybe this happens if the inbound caller hangs up while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there. (The
preceding line of the log is a call from almost an hour earlier). In that
case though I'd expect to see 1-CANCEL in the log instead. Perhaps if
the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to
prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Thanks in advance for any and all advice!
~Brian
--
Brian Henning, Software Engineer
/\Pine Research Instrumentation
//\\ 5908 Triangle Drive
///\\\ Raleigh, NC 27617
USA
||
||phone: 919.782.8320
fax: 919.782.8323
email: bhenn...@pineinst.com
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Re: [asterisk-users] How to debug MixMonitor misbehaviour
Thanks for the input. I think that works as my other recordings work. I will
test that again regardless.
Is there no real other way to know why MixMonitor fails or look more into
it?
Regards,
Bruce
On Wed, May 4, 2011 at 5:03 AM, salaheddine elharit
salah.elharit...@gmail.com wrote:
hi
you can add this in extenssion.conf
exten = 223,1,Answer()
exten = 223,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 223,3,Dial(SIP/223)
exten = 223,4,Hangup()
i can record without any issue in /var/spool/asterisk/monitor
2011/5/4 Bruce B bruceb...@gmail.com
Thanks for the input.
Yes, I did call out many times, but the recording doesn't happen even
after the call is bridged and there is two way audio. I also took out the
b option and so it should recording the ringing right (even before call is
bridged) but it doesn't do that or any recording at all.
Any other suggestions as to what I can do to see why this is not
recording?
Regards,
On Tue, May 3, 2011 at 2:13 AM, virendra bhati virbh...@gmail.comwrote:
Hi,
As per your Dialplan MixMonitor will work after call bridge, In you case
still call is not bridge. That's why MixMonitor is waiting of call bridge...
*
MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)
option b=** A bridge flag allows recording to only take place when the
channel is bridged.*
So just for test make sip call and start mixmonitor to test the recorded
file.
default path od recording id
*
/var/spool/asterisk/monitor/
*
On Tue, May 3, 2011 at 10:40 AM, Bruce B bruceb...@gmail.com wrote:
Hi everyone,
For some reason MixMonitor doesn't record when it should; It actually
shows the MixMonitor line just fine on the CLI. How can MixMonitor be
debugged for things like privilege issues or filename issues?
**I had this working at one point and then stopped working. Not sure
what I changed.
System Info:
Asterisk 1.4.21.2
Queuemetrics 1.6.3.0
[queuedial]
; this piece of dialplan is just a calling hook into the
[qm-queuedial] context that actually does the
; outbound dialing - replace as needed - just fill in the same
variables.
exten = _XXX.,1,Set(QDIALER_QUEUE=q-${EXTEN:0:3})
exten = _XXX.,n,Set(QDIALER_NUMBER=${EXTEN:3})
exten = _XXX.,n,Set(QDIALER_AGENT=Agent/${CALLERID(num)})
exten = _XXX.,n,Set(QDIALER_CHANNEL=ZAP/g0/${QDIALER_NUMBER})
exten = _XXX.,n,Set(QueueName=${QDIALER_QUEUE})
*exten = _XXX.,n,MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)*
exten = _XXX.,n,Goto(qm-queuedial,s,1)
CLI output:
-- Called 4904166356574@queuedial/n
-- Executing [4904166356574@queuedial:1]
Set(Local/4904166356574@queuedial-d851,2, QDIALER_QUEUE=q-490) in
new stack
-- Executing [4904166356574@queuedial:2]
Set(Local/4904166356574@queuedial-d851,2,
QDIALER_NUMBER=4166356574) in new stack
-- Executing [4904166356574@queuedial:3]
Set(Local/4904166356574@queuedial-d851,2,
QDIALER_AGENT=Agent/19053640558) in new stack
-- Executing [4904166356574@queuedial:4]
Set(Local/4904166356574@queuedial-d851,2,
QDIALER_CHANNEL=ZAP/g0/4166356574) in new stack
-- Executing [4904166356574@queuedial:5]
Set(Local/4904166356574@queuedial-d851,2, QueueName=q-490) in new
stack
*-- Executing [4904166356574@queuedial:6]
MixMonitor(Local/4904166356574@queuedial-d851,2,
Q-q-490-1304399098.18.WAV|b|) in new stack*
-- Executing [4904166356574@queuedial:7]
Goto(Local/4904166356574@queuedial-d851,2, qm-queuedial|s|1) in new
stack
-- Goto (qm-queuedial,s,1)
Trying to locate file:
root@pbx:~ $ updatedb
root@pbx:~ $ locate Q-q-490-1304399098.18.WAV
root@pbx:~ $ ls /var/spool/asterisk/monitor/Q-q*
ls: /var/spool/asterisk/monitor/Q-q*: No such file or directory
I also turned on the Debug but I couldn't see anything out of the norm.
As you can see above the CLI output is just fine.
Thanks,
Bruce
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asterisk
Re: [asterisk-users] How to debug MixMonitor misbehaviour
Thanks for the input.
Yes, I did call out many times, but the recording doesn't happen even after
the call is bridged and there is two way audio. I also took out the b
option and so it should recording the ringing right (even before call is
bridged) but it doesn't do that or any recording at all.
Any other suggestions as to what I can do to see why this is not recording?
Regards,
On Tue, May 3, 2011 at 2:13 AM, virendra bhati virbh...@gmail.com wrote:
Hi,
As per your Dialplan MixMonitor will work after call bridge, In you case
still call is not bridge. That's why MixMonitor is waiting of call bridge...
*
MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)
option b=** A bridge flag allows recording to only take place when the
channel is bridged.*
So just for test make sip call and start mixmonitor to test the recorded
file.
default path od recording id
*
/var/spool/asterisk/monitor/
*
On Tue, May 3, 2011 at 10:40 AM, Bruce B bruceb...@gmail.com wrote:
Hi everyone,
For some reason MixMonitor doesn't record when it should; It actually
shows the MixMonitor line just fine on the CLI. How can MixMonitor be
debugged for things like privilege issues or filename issues?
**I had this working at one point and then stopped working. Not sure what
I changed.
System Info:
Asterisk 1.4.21.2
Queuemetrics 1.6.3.0
[queuedial]
; this piece of dialplan is just a calling hook into the [qm-queuedial]
context that actually does the
; outbound dialing - replace as needed - just fill in the same variables.
exten = _XXX.,1,Set(QDIALER_QUEUE=q-${EXTEN:0:3})
exten = _XXX.,n,Set(QDIALER_NUMBER=${EXTEN:3})
exten = _XXX.,n,Set(QDIALER_AGENT=Agent/${CALLERID(num)})
exten = _XXX.,n,Set(QDIALER_CHANNEL=ZAP/g0/${QDIALER_NUMBER})
exten = _XXX.,n,Set(QueueName=${QDIALER_QUEUE})
*exten = _XXX.,n,MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)*
exten = _XXX.,n,Goto(qm-queuedial,s,1)
CLI output:
-- Called 4904166356574@queuedial/n
-- Executing [4904166356574@queuedial:1]
Set(Local/4904166356574@queuedial-d851,2, QDIALER_QUEUE=q-490) in new
stack
-- Executing [4904166356574@queuedial:2]
Set(Local/4904166356574@queuedial-d851,2, QDIALER_NUMBER=4166356574)
in new stack
-- Executing [4904166356574@queuedial:3]
Set(Local/4904166356574@queuedial-d851,2,
QDIALER_AGENT=Agent/19053640558) in new stack
-- Executing [4904166356574@queuedial:4]
Set(Local/4904166356574@queuedial-d851,2,
QDIALER_CHANNEL=ZAP/g0/4166356574) in new stack
-- Executing [4904166356574@queuedial:5]
Set(Local/4904166356574@queuedial-d851,2, QueueName=q-490) in new
stack
*-- Executing [4904166356574@queuedial:6]
MixMonitor(Local/4904166356574@queuedial-d851,2,
Q-q-490-1304399098.18.WAV|b|) in new stack*
-- Executing [4904166356574@queuedial:7]
Goto(Local/4904166356574@queuedial-d851,2, qm-queuedial|s|1) in new
stack
-- Goto (qm-queuedial,s,1)
Trying to locate file:
root@pbx:~ $ updatedb
root@pbx:~ $ locate Q-q-490-1304399098.18.WAV
root@pbx:~ $ ls /var/spool/asterisk/monitor/Q-q*
ls: /var/spool/asterisk/monitor/Q-q*: No such file or directory
I also turned on the Debug but I couldn't see anything out of the norm. As
you can see above the CLI output is just fine.
Thanks,
Bruce
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+91-9172341457
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[asterisk-users] How to debug MixMonitor misbehaviour
Hi everyone,
For some reason MixMonitor doesn't record when it should; It actually shows
the MixMonitor line just fine on the CLI. How can MixMonitor be debugged for
things like privilege issues or filename issues?
**I had this working at one point and then stopped working. Not sure what I
changed.
System Info:
Asterisk 1.4.21.2
Queuemetrics 1.6.3.0
[queuedial]
; this piece of dialplan is just a calling hook into the [qm-queuedial]
context that actually does the
; outbound dialing - replace as needed - just fill in the same variables.
exten = _XXX.,1,Set(QDIALER_QUEUE=q-${EXTEN:0:3})
exten = _XXX.,n,Set(QDIALER_NUMBER=${EXTEN:3})
exten = _XXX.,n,Set(QDIALER_AGENT=Agent/${CALLERID(num)})
exten = _XXX.,n,Set(QDIALER_CHANNEL=ZAP/g0/${QDIALER_NUMBER})
exten = _XXX.,n,Set(QueueName=${QDIALER_QUEUE})
*exten = _XXX.,n,MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)*
exten = _XXX.,n,Goto(qm-queuedial,s,1)
CLI output:
-- Called 4904166356574@queuedial/n
-- Executing [4904166356574@queuedial:1]
Set(Local/4904166356574@queuedial-d851,2, QDIALER_QUEUE=q-490) in new
stack
-- Executing [4904166356574@queuedial:2]
Set(Local/4904166356574@queuedial-d851,2, QDIALER_NUMBER=4166356574) in
new stack
-- Executing [4904166356574@queuedial:3]
Set(Local/4904166356574@queuedial-d851,2,
QDIALER_AGENT=Agent/19053640558) in new stack
-- Executing [4904166356574@queuedial:4]
Set(Local/4904166356574@queuedial-d851,2,
QDIALER_CHANNEL=ZAP/g0/4166356574) in new stack
-- Executing [4904166356574@queuedial:5]
Set(Local/4904166356574@queuedial-d851,2, QueueName=q-490) in new stack
*-- Executing [4904166356574@queuedial:6]
MixMonitor(Local/4904166356574@queuedial-d851,2,
Q-q-490-1304399098.18.WAV|b|) in new stack*
-- Executing [4904166356574@queuedial:7]
Goto(Local/4904166356574@queuedial-d851,2, qm-queuedial|s|1) in new
stack
-- Goto (qm-queuedial,s,1)
Trying to locate file:
root@pbx:~ $ updatedb
root@pbx:~ $ locate Q-q-490-1304399098.18.WAV
root@pbx:~ $ ls /var/spool/asterisk/monitor/Q-q*
ls: /var/spool/asterisk/monitor/Q-q*: No such file or directory
I also turned on the Debug but I couldn't see anything out of the norm. As
you can see above the CLI output is just fine.
Thanks,
Bruce
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[asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?
Hi everyone, How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is there any easy way to simulate a distorted SIP line temporarily for testing? I am appreciate experienced inputs. 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
Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?
Thanks for the input guys. What Tony and Satish suggested are alone the lines of what I need. It gives me a controlled solution. So, I can change the level of distortion as I please. Using tc I pretty much killed the line to the point I wasn't able to receive call and terminal was really slow as well. I am going to try the the packet drop method now. I think that is the right one for the situation. Thanks again On Thu, Apr 28, 2011 at 11:57 AM, Tony Mountifield t...@softins.co.ukwrote: In article banlktim8w+vjjj87oyy1mvppsfwflut...@mail.gmail.com, Bruce B bruceb...@gmail.com wrote: How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is there any easy way to simulate a distorted SIP line temporarily for testing? I am appreciate experienced inputs. You could use iptables to cause random packet loss. See http://code.nomad-labs.com/2010/03/11/simulating-dropped-packets-aka-crappy-internets-with-iptables/ for examples. You might want to precede those rules with ACCEPT rules for the traffic you want to remain reliable (such as TCP connections). Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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
Re: [asterisk-users] Call Center Reporting
Hi Bilal, Probably there is no open source tool or a good ones available. But few of them I worked with provide up to 2 users free of cost license type of reporting. Reporting for Call Centers can get very complicated. Once you explore some of the commercial apps you will notice how extensive they can get. This is specially true if you are replacing an existing commercial system as you client won't want a mickey mouse replacement but rather a full-fledged call center application. To set down and code for it, it will probably take months to match anything commercially available. I suggest you explore your options before coding it or even attempting queue logs into SQL as that is just he beginning of the work and presentation, ***real-time***, graphs, administration portal, and tons more things are needed to make it a complete suite. Not to forget that this will require continuous updates at the pace of Digium changing Asterisk versions (most of the time as dial-plans changes or queue-log events changes, or if AMI events change). Of course it's possible like other posts suggested but is it economical to embark on it for a single small project? I am not sure Just a thought. -Bruce On Mon, Apr 18, 2011 at 9:23 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Mon, Apr 18, 2011 at 8:18 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: If all the details you need to compile your reports can be found in existing databases (Asterisk's CDR database stores the details of calls; you may need to get user login/out events from a separate database), Logging the queue_log to MySQL and then setting up a trigger that inserts/updates data to other tables (such as something like agent_status and call_status), along with the CDR, will allow the OP to get pretty much everything they want. (*OP, if you need something substantially more than the stats I mentioned in my earlier post, definitely feel free to email me with details. That way, not only can I help you, but I can make the open source statistics solution I'm working on even better)* A hint: Do the whole thing -- or as much of it as it takes to prove to yourself that you're on the right track -- by hand first, entering all the queries yourself in the mysql prompt (or phpmyadmin), *before* you try to write a program to do it. You will save yourself much heartache that way. AJ, truer words have not been oft spoken! I'd also add that creating views helps if you have complex queries (just to shorten the query that has to be issued from the end program that gets written). -- _ -- 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
Re: [asterisk-users] Occasional call from asterisk
I wonder if you can test to see if this happens if you had an analogue phone set connected. And if it doesn't then I am wondering why Asterisk or Sangoma card is so sensitive and maybe the sensor can be set a bit higher so these calls don't end-up ringing like they don't if an analogue phone set was connected to the line (at least that was my case). -Bruce On Mon, Apr 11, 2011 at 9:52 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Brian Henning *Sent:* Monday, April 11, 2011 8:47 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Occasional call from asterisk Bruce B said: We experience exact same thing on DAHDI with Sangoma USB FXO device on short circuited lines. Phantom calls are actually due to a short in the lines that happen occasionally. -Bruce Also, Warren Selby said: I've seen this on cases where a phantom call comes in on a DAHDI channel - these calls were the results of faulty wiring on the part of the telco. Check your logs for any errors on your DAHDI channels around the time of the ghost calls. It could also be a case of someone calls in and then hangs up before the call is actually passed to asterisk, and the telco is just slow to hangup the call. H. I do see this in the /var/log/asterisk/messages log: [Apr 5 00:05:36] NOTICE[9579] chan_dahdi.c: Got event 4 (Alarm)... [Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 5 (No more alarm)... [Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 17 (Polarity Reversal)... [Apr 5 00:05:41] WARNING[2400] chan_dahdi.c: Detected alarm on channel 3: Red Alarm [Apr 5 00:05:42] NOTICE[2400] chan_dahdi.c: Alarm cleared on channel 3 [Apr 5 00:05:51] WARNING[2400] chan_dahdi.c: Detected alarm on channel 2: Red Alarm [Apr 5 00:05:53] NOTICE[2400] chan_dahdi.c: Alarm cleared on channel 2 …and it appears to coincide directly with an ‘asterisk’ entry in my SIP phone’s missed call log. Our wiring is sketchy; this is known at our facility. Some years ago a backhoe severed the entire trunk and the repair work was of questionable quality. Also our service entry point / punch-down area is a rat’s nest (one building and service is shared by three companies). I guess I can chalk this behavior up to the wiring. Thanks for the input! Cheers, ~Brian *[Danny Nicholas] * *I’ll add another vote to “shoot the phone company” – our wiring goes to heck whenever it rains and we can expect a few ‘phantom calls” from 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 -- _ -- 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 it the normal behaviore for AGI and DeadAGI to terminate AFTER the h extension?
Thanks for the input but I am not sure if that answer my question of if it's
normal behaviour for AGI scrip to terminate after the h extension rather
than end of x extension even if it was only run in x extension.
Regards,
On Mon, Apr 11, 2011 at 6:34 AM, Pezhman Lali l...@lopl.net wrote:
Dear
there is some problem.
the true way for running php script, is using agi not system.
second after 5 sec, a lot of channel variables were removed, it makes your
program wrong.
with some little experience you can add your script to a2billing, try it.
best
On Sat, Apr 9, 2011 at 7:22 PM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
Trying to run a php script after DeadAGI for A2Billing does it's magic.
This is the dialplan:
[a2billing]
exten = _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN}
${UNIQUEID})
exten = _X.,n,AGI(a2billing.php,1)
exten = _X.,n,Hangup()
*exten = h,1,Wait(5)*
*exten = h,n,System(php post-call.php ${CALLERID(num)} ${UNIQUEID})*
As you can see above, I even added a 5 seconds wait time before running my
post-call script but the problem is that DeadAGI and AGI actually terminates
after the h,n extension is run. I thought this weired because X
extension is already finished at X,3.
Is this normal behavioral for AGI/DeadAGI?
Is there a workaround this other than including my post-call script into
the a2billing.php itself?
Output from CLI for proof:
-- Executing [h@a2billing:1] Wait(SIP/101-0034, 5) in new stack
*-- Executing [h@a2billing:2]* System(SIP/101-0034, php
post-call.php 101 1302360230.52) in new stack
-*- SIP/101-0034AGI Script a2billing.php completed, returning
-1*
*
*
Thanks,
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Re: [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the h extension?
You probably didn't read over my originally post carefully. In the dialplan
A2billing.php script is called in the X extension. Then there is
X,n,Hangup() so now X extension is dead.
After that in h extension I have ANOTHER script running. However, the CLI
output (which again I posted in my original post) shows that the
A2Billing.php script which was run in the X extension actually terminated
at the end of h extension.
I would appreciate it if someone with knowledge can please weigh in.
Regards,
On Mon, Apr 11, 2011 at 12:06 PM, Pezhman Lali l...@lopl.net wrote:
h is hangup extension, and will be executed after hangup
On Mon, Apr 11, 2011 at 6:36 PM, Bruce B bruceb...@gmail.com wrote:
Thanks for the input but I am not sure if that answer my question of if
it's normal behaviour for AGI scrip to terminate after the h extension
rather than end of x extension even if it was only run in x extension.
Regards,
On Mon, Apr 11, 2011 at 6:34 AM, Pezhman Lali l...@lopl.net wrote:
Dear
there is some problem.
the true way for running php script, is using agi not system.
second after 5 sec, a lot of channel variables were removed, it makes
your program wrong.
with some little experience you can add your script to a2billing, try it.
best
On Sat, Apr 9, 2011 at 7:22 PM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
Trying to run a php script after DeadAGI for A2Billing does it's magic.
This is the dialplan:
[a2billing]
exten = _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN}
${UNIQUEID})
exten = _X.,n,AGI(a2billing.php,1)
exten = _X.,n,Hangup()
*exten = h,1,Wait(5)*
*exten = h,n,System(php post-call.php ${CALLERID(num)} ${UNIQUEID})*
As you can see above, I even added a 5 seconds wait time before running
my post-call script but the problem is that DeadAGI and AGI actually
terminates after the h,n extension is run. I thought this weired because
X extension is already finished at X,3.
Is this normal behavioral for AGI/DeadAGI?
Is there a workaround this other than including my post-call script into
the a2billing.php itself?
Output from CLI for proof:
-- Executing [h@a2billing:1] Wait(SIP/101-0034, 5) in new
stack
*-- Executing [h@a2billing:2]* System(SIP/101-0034, php
post-call.php 101 1302360230.52) in new stack
-*- SIP/101-0034AGI Script a2billing.php completed, returning
-1*
*
*
Thanks,
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