[asterisk-users] Asterisk 11.14.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.14.0 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-24348 - Built-in editline tab complete segfault with MALLOC_DEBUG (Reported by Walter Doekes) * ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls (Reported by Torrey Searle) * ASTERISK-23768 - [patch] Asterisk man page contains a (new) unquoted minus sign (Reported by Jeremy Lainé) * ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits (Reported by Jeremy Lainé) * ASTERISK-20567 - bashism in autosupport (Reported by Tzafrir Cohen) * ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with realtime peers (Reported by ibercom) * ASTERISK-24384 - chan_motif: format capabilities leak on module load error (Reported by Corey Farrell) * ASTERISK-24385 - chan_sip: process_sdp leaks on an error path (Reported by Corey Farrell) * ASTERISK-24378 - Release AMI connections on shutdown (Reported by Corey Farrell) * ASTERISK-24354 - AMI sendMessage closes AMI connection on error (Reported by Peter Katzmann) * ASTERISK-24390 - astobj2: REF_DEBUG reports false leaks with ao2_callback with OBJ_MULTIPLE (Reported by Corey Farrell) * ASTERISK-24326 - res_rtp_asterisk: ICE-TCP candidates are incorrectly attempted (Reported by Joshua Colp) * ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM (Reported by Michael Myles) * ASTERISK-24383 - res_rtp_asterisk: Crash if no candidates received for component (Reported by Kevin Harwell) * ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak (Reported by NITESH BANSAL) * ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak (Reported by Torrey Searle) * ASTERISK-24406 - Some caller ID strings are parsed differently since 11.13.0 (Reported by Etienne Lessard) * ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30 (Reported by Tzafrir Cohen) * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by Tzafrir Cohen) * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE (Reported by Paolo Compagnini) * ASTERISK-18923 - res_fax_spandsp usage counter is wrong (Reported by Grigoriy Puzankin) * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by Corey Farrell) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-23846 - Unistim multilines. Loss of voice after second call drops (on a second line). (Reported by Rustam Khankishyiev) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 (Reported by Patrick Laimbock) * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing leak (Reported by Corey Farrell) * ASTERISK-24430 - missing letter "p" in word response in OriginateResponse event documentation (Reported by Dafi Ni) * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by Corey Farrell) * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) * ASTERISK-24304 - asterisk crashing randomly because of unistim channel (Reported by dhanapathy sathya) * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by Nick Adams) * ASTERISK-24466 - app_queue: fix a couple leaks to struct call_queue (Reported by Corey Farrell) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24307 - Unintentional memory retention in stringfields (Reported by Etienne Lessard) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.14.0 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.asteri
[asterisk-users] Asterisk 12.7.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 12.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.7.0 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-24339 - Swagger API Docs have incorrect basePath (Reported by Bradley Watkins) * ASTERISK-24348 - Built-in editline tab complete segfault with MALLOC_DEBUG (Reported by Walter Doekes) * ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls (Reported by Torrey Searle) * ASTERISK-24295 - crash: creating out of dialog OPTIONS request crashes (Reported by Rogger Padilla) * ASTERISK-23768 - [patch] Asterisk man page contains a (new) unquoted minus sign (Reported by Jeremy Lainé) * ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits (Reported by Jeremy Lainé) * ASTERISK-20567 - bashism in autosupport (Reported by Tzafrir Cohen) * ASTERISK-24350 - PJSIP shows commands prints unneeded headers (Reported by snuffy) * ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with realtime peers (Reported by ibercom) * ASTERISK-24362 - res_hep leaks reference to configuration (Reported by Corey Farrell) * ASTERISK-23781 - outgoing missing as enum from contrib/ast-db-manage/config (Reported by Stephen More) * ASTERISK-24199 - 'ALL' is specified in pjsip.conf.sample for TLS cipher but it is not valid (Reported by Joshua Colp) * ASTERISK-24262 - AMI CoreShowChannel missing several output fields and event documentation (Reported by Mitch Claborn) * ASTERISK-24356 - PJSIP: Directed pickup causes deadlock (Reported by Richard Mudgett) * ASTERISK-24195 - bridge_native_rtp: Removing mixmonitor from a native RTP capable smart bridge doesn't cause the bridge to resume being a native rtp bridge (Reported by Jonathan Rose) * ASTERISK-24384 - chan_motif: format capabilities leak on module load error (Reported by Corey Farrell) * ASTERISK-24385 - chan_sip: process_sdp leaks on an error path (Reported by Corey Farrell) * ASTERISK-24378 - Release AMI connections on shutdown (Reported by Corey Farrell) * ASTERISK-24369 - res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations (Reported by Matt Jordan) * ASTERISK-24382 - chan_pjsip: Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel results in an invalid reference of a channel pvt and a FRACK (Reported by Matt Jordan) * ASTERISK-24370 - res_pjsip/pjsip_options: OPTIONS request sent to Asterisk with no user in request is always 404'd (Reported by Matt Jordan) * ASTERISK-24224 - When using Bridge() dialplan application, surrogate channel appears in list and call count is inflated. (Reported by Mark Michelson) * ASTERISK-24354 - AMI sendMessage closes AMI connection on error (Reported by Peter Katzmann) * ASTERISK-24398 - Initialize auth_rejection_permanent on client state to the configuration parameter value (Reported by Matt Jordan) * ASTERISK-24326 - res_rtp_asterisk: ICE-TCP candidates are incorrectly attempted (Reported by Joshua Colp) * ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM (Reported by Michael Myles) * ASTERISK-24383 - res_rtp_asterisk: Crash if no candidates received for component (Reported by Kevin Harwell) * ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak (Reported by NITESH BANSAL) * ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak (Reported by Torrey Searle) * ASTERISK-24387 - res_pjsip: rport sent from UAS MUST include the port that the UAC sent the request on (Reported by Matt Jordan) * ASTERISK-24406 - Some caller ID strings are parsed differently since 11.13.0 (Reported by Etienne Lessard) * ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30 (Reported by Tzafrir Cohen) * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by Tzafrir Cohen) * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE (Reported by Paolo Compagnini) * ASTERISK-18923 - res_fax_spandsp usage counter is wrong (Reported by Grigoriy Puzankin) * ASTERISK-24394 - CDR: FRACK with PJSIP directed pickup. (Reported by Richard Mudgett) * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by Corey Farrell) * ASTERISK-24321 - SIP deadlock when running automated queues tests (Reported by Steve P
[asterisk-users] Asterisk 1.8.32.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.32.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.32.0 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-24348 - Built-in editline tab complete segfault with MALLOC_DEBUG (Reported by Walter Doekes) * ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls (Reported by Torrey Searle) * ASTERISK-23768 - [patch] Asterisk man page contains a (new) unquoted minus sign (Reported by Jeremy Lainé) * ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits (Reported by Jeremy Lainé) * ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with realtime peers (Reported by ibercom) * ASTERISK-24390 - astobj2: REF_DEBUG reports false leaks with ao2_callback with OBJ_MULTIPLE (Reported by Corey Farrell) * ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM (Reported by Michael Myles) * ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak (Reported by NITESH BANSAL) * ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak (Reported by Torrey Searle) * ASTERISK-24406 - Some caller ID strings are parsed differently since 11.13.0 (Reported by Etienne Lessard) * ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30 (Reported by Tzafrir Cohen) * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by Tzafrir Cohen) * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE (Reported by Paolo Compagnini) * ASTERISK-18923 - res_fax_spandsp usage counter is wrong (Reported by Grigoriy Puzankin) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 (Reported by Patrick Laimbock) * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by Nick Adams) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24307 - Unintentional memory retention in stringfields (Reported by Etienne Lessard) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.32.0 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
Re: [asterisk-users] Webinar Gratuíto, Como evitar fraudes em telefonia
2014-11-10 9:53 GMT-06:00 Flavio Goncalves : > A SipPulse acaba de liberar o TFPS (www.tfps.co), solução para combate a > fraudes de fomento de tráfego internacional em telefonia. O sistema é capaz > de detectar 99.99% das tentaivas de fraude em tempo real. > > Durante o Webinar, abordaremos como proteger servidores Asterisk e > Elastix/FreePBX de fraudes, medidas basicas como configuração de firewall e > remoção de serviços desnecessários e em seguida como configurar os serviços > de prevenção à fraudes. > > Registre-se no evento > em:https://voffice.webex.com/voffice/onstage/g.php?MTID=e1bc7fec6e11b4d0114accc8cf20b36e6 > > Flavio E. Goncalves > SipPulse > Please don't advertise on the asterisk-users mailing list. Advertisements for products and services belong on the asterisk-biz mailing list [1]. Matt [1] http://lists.digium.com/mailman/listinfo/asterisk-biz -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Webinar Gratuíto, Como evitar fraudes em telefonia
A SipPulse acaba de liberar o TFPS (www.tfps.co), solução para combate a fraudes de fomento de tráfego internacional em telefonia. O sistema é capaz de detectar 99.99% das tentaivas de fraude em tempo real. Durante o Webinar, abordaremos como proteger servidores Asterisk e Elastix/FreePBX de fraudes, medidas basicas como configuração de firewall e remoção de serviços desnecessários e em seguida como configurar os serviços de prevenção à fraudes. Registre-se no evento em: https://voffice.webex.com/voffice/onstage/g.php?MTID=e1bc7fec6e11b4d0114accc8cf20b36e6 Flavio E. Goncalves SipPulse -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Function to get mailbox for a PJSIP Endpoint?
Thank you George, That works.
If anyone is curious here is what I'm doing for my mail checking extension.
exten => *98,1,NoOP()
same => n,Answer()
same => n,Set(ENDPOINT=${CHANNEL(endpoint)}) ; Get the peer
same => n,Set(MAILBOX=${PJSIP_ENDPOINT(${ENDPOINT},mailboxes)})
same => n,ExecIf($[${ISNULL(${MAILBOX})} =
1]?Set(MAILBOX=${AST_SORCERY(res_pjsip,aor,${ENDPOINT},mailboxes)}))
same => n,ExecIf($[${ISNULL(${MAILBOX})} =
0]?VoicemailMain(${MAILBOX},s))
same => n,ExecIf($[${ISNULL(${MAILBOXCONTEXT})} =
0]?VoicemailMain(@${MAILBOXCONTEXT}))
same => n,Playtones(congestion)
same => n,Congestion(8)
same => n,Hangup()
On Fri, Nov 7, 2014 at 6:26 PM, George Joseph
wrote:
>
>
> On Fri, Nov 7, 2014 at 6:20 AM, Joshua Colp wrote:
>
>> John Kiniston wrote:
>>
>>> Here's my config, I am configuring the mailboxes as you see below in the
>>> aor.
>>>
>>> It looked like that was the recommended place to configure it?
>>>
>>
>> There's no recommendation one way or the other really - it depends on
>> what the devices in question are expecting.
>>
>> If I configure the mailboxes in the Endpoint then the PBX will be
>>> sending unsolicited MWI vs if I configure them in the AOR then my phones
>>> subscribe for MWI Correct?
>>>
>>
>> Correct. Since you are configuring on the AOR there is no current way to
>> grab that information in the dialplan from the AOR.
>>
>>
> Actually you can use the AST_SORCERY dialplan function function to
> retrieve any variable from any PJSIP object. AST_SORCERY("res_pjsip",
> "aor", "myaor", "mailboxes")
>
>
>>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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Re: [asterisk-users] Subscribe event "ua-profile"
Steven Howes wrote: On 10 Nov 2014, at 13:01, Norman Laidla wrote: Well, pants. It actually is causing a problem, because the phone doesn't use any other methods to register to Asterisk. This is a bit of a big issue. It's not really using ua-profile to register. It's using that to get profile information so it can provision itself and become aware if its configuration changes. What softphone is it? It sounds like rather odd behaviour. It's odd to me that this is a required RFC. Is this softphone written to be used against a specific platform? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
Re: [asterisk-users] Subscribe event "ua-profile"
On 10 Nov 2014, at 13:01, Norman Laidla wrote: > Well, pants. It actually is causing a problem, because the phone doesn't use > any other methods to register to Asterisk. This is a bit of a big issue. What softphone is it? It sounds like rather odd behaviour. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Subscribe event "ua-profile"
Well, pants. It actually is causing a problem, because the phone doesn't use any other methods to register to Asterisk. This is a bit of a big issue. Hopefully support for the event will be implemented somewhere down the line. Unfortunately, I lack the skills to do it myself. Best regards, Norman From: asterisk-users-boun...@lists.digium.com on behalf of Joshua Colp Sent: Monday, November 10, 2014 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Subscribe event "ua-profile" Norman Laidla wrote: > Morning! Kia ora, > I'm trying to subscribe a softphone to an Asterisk 11 server, but it > sends an "ua-profile" event that Asterisk immediately rejects with a 489 > Bad Event error. Is this event not supported at all? Are there any > workarounds? That's correct, there is no support for ua-profile and a 489 is the correct response. This shouldn't cause any problems though, unless the softphone absolutely requires that RFC to be implemented in which case it won't work. The only workaround for likely be to implement it if that is the case. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
Re: [asterisk-users] Subscribe event "ua-profile"
Norman Laidla wrote: Morning! Kia ora, I'm trying to subscribe a softphone to an Asterisk 11 server, but it sends an "ua-profile" event that Asterisk immediately rejects with a 489 Bad Event error. Is this event not supported at all? Are there any workarounds? That's correct, there is no support for ua-profile and a 489 is the correct response. This shouldn't cause any problems though, unless the softphone absolutely requires that RFC to be implemented in which case it won't work. The only workaround for likely be to implement it if that is the case. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
[asterisk-users] Subscribe event "ua-profile"
Morning! I'm trying to subscribe a softphone to an Asterisk 11 server, but it sends an "ua-profile" event that Asterisk immediately rejects with a 489 Bad Event error. Is this event not supported at all? Are there any workarounds? Best regards, Norman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
