Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers
On Fri, Feb 6, 2015 at 5:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 6 February 2015 at 07:54, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, Got a question regarding custom announcements in Asterisk. My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? Keep up the good work! cheers, Olli -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi All of that is possible and is exactly what we do, both for customer sounds and for call recordings. Just make sure you have resilience in your shared storage device. Alternatively, you could use something like Puppet to deploy the files to all the servers. This is basically what we do, we use puppet to help distribute files to remote servers while still using app_queue. Shared network drive also works. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.16.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.16.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.16.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: --- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS from JSSIP (Reported by Badalian Vyacheslav) * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag enabled (Reported by Richard Mudgett) * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is enabled (Reported by Andreas Steinmetz) * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int (Reported by Walter Doekes) * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped' (Reported by Rusty Newton) * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present on startup (Reported by Richard Kenner) * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment) (Reported by Karsten Wemheuer) * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT (Reported by Kristian Høgh) * ASTERISK-20744 - [patch] Security event logging does not work over syslog (Reported by Michael Keuter) * ASTERISK-23850 - Park Application does not respect Return Context Priority (Reported by Andrew Nagy) * ASTERISK-23991 - [patch]asterisk.pc file contains a small error in the CFlags returned (Reported by Diederik de Groot) * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup (Reported by LEI FU) * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit hosts (Reported by Ben Klang) * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event (Reported by Gareth Palmer) * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser' (Reported by HZMI8gkCvPpom0tM) * ASTERISK-24719 - ConfBridge recording channels get stuck when recording started/stopped more than once (Reported by Richard Mudgett) * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported by Kevin Harwell) * ASTERISK-24728 - tcptls: Bad file descriptor error when reloading chan_sip (Reported by Kevin Harwell) * ASTERISK-24676 - Security Vulnerability: URL request injection in libCURL (CVE-2014-8150) (Reported by Matt Jordan) * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL versions (Reported by Jared Biel) * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by Stephan Eisvogel) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0 Thank you for your continued support of 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
[asterisk-users] Asterisk 13.2.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: --- * ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them all at the same time. (Reported by Richard Mudgett) * ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard (Reported by Kevin Harwell) * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS from JSSIP (Reported by Badalian Vyacheslav) * ASTERISK-24607 - res_pjsip_session: re-INVITE with declined media streams results in 488 (Reported by Matt Jordan) * ASTERISK-24563 - Direct Media calls within private network sometimes get one way audio (Reported by Kevin Harwell) * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core (Reported by Matt Jordan) * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag enabled (Reported by Richard Mudgett) * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is enabled (Reported by Andreas Steinmetz) * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int (Reported by Walter Doekes) * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra channel (Reported by Niklas Larsson) * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible (Reported by Yaniv Simhi) * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped' (Reported by Rusty Newton) * ASTERISK-24513 - Local channel apparently leaked in off-nominal DTMF attended transfer (Reported by Mark Michelson) * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present on startup (Reported by Richard Kenner) * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment) (Reported by Karsten Wemheuer) * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall calls to the transferrer. (Reported by Richard Mudgett) * ASTERISK-24376 - res_pjsip_refer: REFER request for remote session attempts to direct channel to external_replaces extension instead of context, without providing for the Referred-To SIP URI (Reported by Matt Jordan) * ASTERISK-24591 - Stasis() side of an ARI originated channel cannot be Redirected (Reported by Kinsey Moore) * ASTERISK-24049 - Asterisk Manager Interface: A number of list type responses aren't using astman_send_listack (Reported by Jonathan Rose) * ASTERISK-24637 - Channel re-enters Stasis() when it should not (Reported by John Bigelow) * ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does not function (Reported by John Kiniston) * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT (Reported by Kristian Høgh) * ASTERISK-20744 - [patch] Security event logging does not work over syslog (Reported by Michael Keuter) * ASTERISK-24665 - Configure check required for pjsip_get_dest_info() (Reported by Mark Michelson) * ASTERISK-23850 - Park Application does not respect Return Context Priority (Reported by Andrew Nagy) * ASTERISK-23991 - [patch]asterisk.pc file contains a small error in the CFlags returned (Reported by Diederik de Groot) * ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown while attempting to publish (Reported by Kevin Harwell) * ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown (Reported by Corey Farrell) * ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails on cross compilation (Reported by abelbeck) * ASTERISK-24624 - Transfer to invalid extension results in hung channel. (Reported by Zane Conkle) * ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf, Incorrect External Addresses is Used in SIP Packets When Responding to INVITE (Reported by David Justl) * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup (Reported by LEI FU) * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit hosts (Reported by Ben Klang) * ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding to most traffic, potential deadlock (Reported by Jeff Collell) * ASTERISK-24560 - Creating a named ARI bridge twice causes a crash (Reported by Kinsey Moore) *
Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers
On 6 February 2015 at 07:54, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, Got a question regarding custom announcements in Asterisk. My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? Keep up the good work! cheers, Olli -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi All of that is possible and is exactly what we do, both for customer sounds and for call recordings. Just make sure you have resilience in your shared storage device. Alternatively, you could use something like Puppet to deploy the files to all the servers. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers
Oops, quite right, how typoful of me! Thanks for the excellent points, I'll look into gluster and puppet and see may way onwards from there. cheers, Olli 2015-02-06 12:32 GMT+02:00 Gareth Blades mailinglist+aster...@dns99.co.uk: On 06/02/15 07:54, Olli Heiskanen wrote: My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? I assume you mean NFS. Yes you can do that although using NFS you will then have a single point of failure and in the standard NFS client configuration if you try to access a file which is on NFS but it is unavailable then the file access will hang. So you might be better off having the files copied onto each of the asterisks servers local file storage or use a redundant file system such as gluster. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Question regarding custom announcements used by several Asterisk servers
On 06/02/15 07:54, Olli Heiskanen wrote: My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? I assume you mean NFS. Yes you can do that although using NFS you will then have a single point of failure and in the standard NFS client configuration if you try to access a file which is on NFS but it is unavailable then the file access will hang. So you might be better off having the files copied onto each of the asterisks servers local file storage or use a redundant file system such as gluster. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] constantly increasing load in Asterisk 11.14
Hi, On Thu, Feb 5, 2015 at 4:56 PM, Scott Griepentrog sgriepent...@digium.com wrote: Can you tell me if the memory usage by Asterisk is also increasing with load over time? Yes, the memory usage does rise a bit. USER PID %CPU %MEMVSZ RSS TTY STAT START TIME COMMAND 11.14 root 26047 8.9 5.0 1302036 407640 ? Sl Jan28 1201:26 asterisk -vvvgf 11.6 root 36336 9.9 1.1 881684 90360 ?Sl Jan28 1328:19 asterisk -vvvgf Interestingly, the CPU graphs show no visible increase. If you need any more information, just let me know. Gareth, the slow increase of load over months was there with 11.5 or 11.6 already, but I can live with a restart every couple month. Once every week is too much. Here is another overview of the load of one machine: http://pbrd.co/1zeFwBy You can see, the Asterisk was running from April to September without much change in load, then was restarted. In November, we updated to 11.14, and from that time, it looks a bit different (and Asterisk needed a lot more restarts). Best Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] PSJIP Leak handle
I have an Asterisk 13 that only processes app Transfer with PJSIP, to the tune of 60 per second. No voice calls. After like 2 hours, I can no longer get into Asterisk. This command, asterisk -r, fails, and also asterisk -rx core show channels, etc. I am returned to the bash prompt. I checked the handles and lsof | grep asterisk |wc -l 7098126 I think there is a kind of handle leak here. Nothing else runs in the box If there is a way to find out what happens, let me know. The dialplan is confidential, for it belongs to my customer,but I can give you access to the box. In short , the app receives a call, checks the number against a database and calls app_transfer. That is it. This is what I see when the command fails: asterisk -r Asterisk SVN-branch-13-r431555M, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = [root@centos7 /]# this command shows the issue, thousands of lines lsof | grep asterisk asterisk 4077 root *450w FIFO 0,8 0t0 110430221 pipe asterisk 4077 root *451r FIFO 0,8 0t0 110429239 pipe asterisk 4077 root *452w FIFO 0,8 0t0 110429239 pipe asterisk 4077 root *453r FIFO 0,8 0t0 110417598 pipe asterisk 4077 root *454w FIFO 0,8 0t0 110417598 pipe asterisk 4077 root *455r FIFO 0,8 0t0 110426507 pipe asterisk 4077 root *456w FIFO 0,8 0t0 110426507 pipe^ It looks like https://issues.asterisk.org/jira/browse/ASTERISK-823 but in fact I am using PJSIP. It is definitely PJSIP, for I replaced the dialplan with plain SIP, and there is no issue, ceteris paribus. Note: I am not a developer and have no idea how to troubleshoot this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
