Re: [asterisk-users] 1.8.32.3 - billsec field does not not increment after call answer - what triggers it? (Joshua Colp)
Hi Joshua Thanks for the response. Interesting that you mention that toll-free numbers can do this, this problem trunk happens to receive calls from the national telecoms provider here (Telkom SA) sourced from a toll-free number. The SIP trunk provider has ported that toll free state telecoms company number for us to a local trunk number with which we authenticate as a SIP peer. (Not sure if what Telkom here in South Africa define as "toll free" on a technical level match what you mention as "toll free" on an American / United States model.) The thing is the number stays in the unanswered state (while in fact answered and working perfectly) for the entire duration of the call, so then it appears 200 OK is not received for the invite. I therefore assume that the actual RTP negotiation process does NOT need the 200 OK for the invite, for the call itself is fine - it is just that Asterisk never "realises", as regards the CDR, that the call was in fact answered. At least it is working and audio flows back and forth. I'll see if I can come up with a SIP trace. Thank you! --- Stefan Viljoen wrote: > > Only this one trunk consistenly has this problem for all calls received over > it. The trunk provider is using sippy on their side. > > What setting / config option for the particular SIP "problem trunk" have my > trunk provider changed on their side to stop Asterisk from recognising that > a call has been answered when it comes in over that trunk? > > It appears some SIP traffic is not being sent by them (or not received by my > Asterisk) that indicates to it a call has been ANSWERED and that it must > start the billsec timer? I can't really speak for the provider but some numbers will stay in inband progress (unanswered) for a bit. Some toll-frees for example. The specific SIP message that would show it as answered would be a 200 OK to the INVITE we sent though. If you provided the SIP log then we could see. Cheers, -- Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] VoiceMail - Allow * for only some users
Hi John,
Ah ha! Excellent. That works.
Now for a further tweak, in my stdexten I set voicemail_option with
with b or u, as appropriate and use ${voicemail_option) instead of
option in the call to Voicemail below so the correct prompt is used.
Thank you!
On Thu, 2016-07-21 at 14:53 -0700, John Kiniston wrote:
> I think you almost have it.
>
> In your vmfwd context have a wildcard match that sends the caller
> back to the originating voicemail and then define specific extensions
> that are allowed to forward.
>
>
> [vmfwd]
> exten => _,1,Voicemail(box@context,option)
> same => n,Hangup
>
> ; Andrew Ruthven
> exten => 7231,1,Set(CALLERID(number)=yyy)
> same => n,Goto(pstn,xxx,1)
>
> On Thu, Jul 21, 2016 at 2:23 PM, Andrew Ruthven yst.net.nz> wrote:
> > Hey,
> >
> > I have free calling to between DDIs and cellphones on our group
> > plan. I
> > figure it'd be nice to allow staff with those cellphones to be able
> > to
> > forward callers to their VoiceMail to their cellphones using the *
> > feature.
> >
> > I have a standard extension macro that has VoiceMail support.
> > So far I've done this by duplicating the standard extension macro,
> > and
> > adding this rule (where ARG1 is the extension):
> >
> > exten => a,1,Goto(vmfwd,${ARG1},1)
> >
> > Then in the vmfwd context I have rules like this (I need to set the
> > CALLERID(number) so our SIP provider accepts the call):
> >
> > ; Andrew Ruthven
> > exten => 7231,1,Set(CALLERID(number)=yyy)
> > exten => 7231,n,Goto(pstn,xxx,1)
> >
> > Which is working nicely. But, I thought, can I simplify this and
> > just
> > have one macro?
> >
> > So I've tried doing the following to fold it into my standard
> > extension
> > macro:
> >
> > 1) Tried using a/_7231 but that didn't match (well, it was worth a
> > try)
> > 2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my
> > extension,
> > but if I disable the 7231 rules in vmfwd, I get:
> >
> > [2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646
> > __ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to
> > invalid
> > extension but no invalid handler:
> > context,exten,priority=vmfwd,7231,1
> >
> > and the call hangs up, not a very nice user experience.
> >
> > The second option could work, as long as the user lands back into
> > VoiceMail if there is no valid extension. I thought about using
> > GoSub,
> > but how do I get the caller back into VoiceMail?
> >
> > I've done a bunch of searching for this, but haven't found any
> > general
> > solutions. Is it possible to do what I'm trying to achieve, or is
> > there
> > a better approach?
> >
> > This is Asterisk 11.13.
> >
> > Cheers,
> > Andrew
> >
> > --
> >
> > Andrew Ruthven, Wellington, New Zealand
> > MIITP, CITPNZ
> >
> > At work: andrew.ruth...@catalyst.net.nz
> > At home: and...@etc.gen.nz
> > Card : http://qr.catalyst.net.nz/907675e1
> > Cloud : NZs only real cloud - https://catalyst.net.nz/cloud
> > GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863
> > LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.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
>
>
>
--
Andrew Ruthven, Wellington, New Zealand
MIITP, CITPNZ
At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Card : http://qr.catalyst.net.nz/907675e1
Cloud : NZs only real cloud - https://catalyst.net.nz/cloud
GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863
LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.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] Asterisk 13 High CPU usage
On Thu, Jul 21, 2016 at 6:02 PM, Chirag Desai wrote: > Hi all, > > I was using 13.5 but upgraded today to 13.9 (13.10 came out a few hours > after I upgraded). > > On both 13.5 and 13.9 asterisk seems to use 100% of the CPU. This usually > happens a few hours after starting asterisk. A restart of asterisk gets the > CPU back down, but only for a little while. > > There asterisk box has no call traffic flowing through it, just 15 or so > registrations. > > I'm sure this is not best practise but for now I am using chan_sip and > pjsip at the same time. My pjsip endpoints are using TLS. > > I am not sure where to start looking in order to debug the CPU usage by > asterisk and would very much appreciate some guidance. > Actually v13.10 has some changes to address high CPU usage in regards to pjsip. Also you should look here for more information: http://blogs.asterisk.org/2016/07/13/asterisk-task-processor-queue-size-warnings/ 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
[asterisk-users] Asterisk 13 High CPU usage
Hi all, I was using 13.5 but upgraded today to 13.9 (13.10 came out a few hours after I upgraded). On both 13.5 and 13.9 asterisk seems to use 100% of the CPU. This usually happens a few hours after starting asterisk. A restart of asterisk gets the CPU back down, but only for a little while. There asterisk box has no call traffic flowing through it, just 15 or so registrations. I'm sure this is not best practise but for now I am using chan_sip and pjsip at the same time. My pjsip endpoints are using TLS. I am not sure where to start looking in order to debug the CPU usage by asterisk and would very much appreciate some guidance. Kind regards, Chirag -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] VoiceMail - Allow * for only some users
I think you almost have it.
In your vmfwd context have a wildcard match that sends the caller back to
the originating voicemail and then define specific extensions that are
allowed to forward.
[vmfwd]
exten => _,1,Voicemail(box@context,option)
same => n,Hangup
; Andrew Ruthven
exten => 7231,1,Set(CALLERID(number)=yyy)
same => n,Goto(pstn,xxx,1)
On Thu, Jul 21, 2016 at 2:23 PM, Andrew Ruthven <
andrew.ruth...@catalyst.net.nz> wrote:
> Hey,
>
> I have free calling to between DDIs and cellphones on our group plan. I
> figure it'd be nice to allow staff with those cellphones to be able to
> forward callers to their VoiceMail to their cellphones using the *
> feature.
>
> I have a standard extension macro that has VoiceMail support.
> So far I've done this by duplicating the standard extension macro, and
> adding this rule (where ARG1 is the extension):
>
> exten => a,1,Goto(vmfwd,${ARG1},1)
>
> Then in the vmfwd context I have rules like this (I need to set the
> CALLERID(number) so our SIP provider accepts the call):
>
> ; Andrew Ruthven
> exten => 7231,1,Set(CALLERID(number)=yyy)
> exten => 7231,n,Goto(pstn,xxx,1)
>
> Which is working nicely. But, I thought, can I simplify this and just
> have one macro?
>
> So I've tried doing the following to fold it into my standard extension
> macro:
>
> 1) Tried using a/_7231 but that didn't match (well, it was worth a try)
> 2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my extension,
> but if I disable the 7231 rules in vmfwd, I get:
>
> [2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646
> __ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to invalid
> extension but no invalid handler: context,exten,priority=vmfwd,7231,1
>
> and the call hangs up, not a very nice user experience.
>
> The second option could work, as long as the user lands back into
> VoiceMail if there is no valid extension. I thought about using GoSub,
> but how do I get the caller back into VoiceMail?
>
> I've done a bunch of searching for this, but haven't found any general
> solutions. Is it possible to do what I'm trying to achieve, or is there
> a better approach?
>
> This is Asterisk 11.13.
>
> Cheers,
> Andrew
>
> --
>
> Andrew Ruthven, Wellington, New Zealand
> MIITP, CITPNZ
>
> At work: andrew.ruth...@catalyst.net.nz
> At home: and...@etc.gen.nz
> Card : http://qr.catalyst.net.nz/907675e1
> Cloud : NZs only real cloud - https://catalyst.net.nz/cloud
> GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863
> LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.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
--
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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every 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 13.10.0 Now Available
On Thu, Jul 21, 2016 at 4:18 PM, Teijo wrote: > > > 21.7.2016, 20:38, Asterisk Development Team kirjoitti: >> >> Bugs fixed in this release: >> --- >> * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. >> (Reported by Alexander Traud) > > > Now it's possible to use dtls_cipher settings such like: > > dtls_cipher=ALL:!SSLv3 > or > dtls_cipher=HIGH:!SSLv3 > > Thank you! > I'll echo that sentiment - Alexander has done a lot of work recently to improve Asterisk's support of available ciphers both in DTLS and SRTP. Thanks Alexander! -- Matthew Jordan Digium, Inc. | CTO 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] VoiceMail - Allow * for only some users
Hey,
I have free calling to between DDIs and cellphones on our group plan. I
figure it'd be nice to allow staff with those cellphones to be able to
forward callers to their VoiceMail to their cellphones using the *
feature.
I have a standard extension macro that has VoiceMail support.
So far I've done this by duplicating the standard extension macro, and
adding this rule (where ARG1 is the extension):
exten => a,1,Goto(vmfwd,${ARG1},1)
Then in the vmfwd context I have rules like this (I need to set the
CALLERID(number) so our SIP provider accepts the call):
; Andrew Ruthven
exten => 7231,1,Set(CALLERID(number)=yyy)
exten => 7231,n,Goto(pstn,xxx,1)
Which is working nicely. But, I thought, can I simplify this and just
have one macro?
So I've tried doing the following to fold it into my standard extension
macro:
1) Tried using a/_7231 but that didn't match (well, it was worth a try)
2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my extension,
but if I disable the 7231 rules in vmfwd, I get:
[2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646
__ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to invalid
extension but no invalid handler: context,exten,priority=vmfwd,7231,1
and the call hangs up, not a very nice user experience.
The second option could work, as long as the user lands back into
VoiceMail if there is no valid extension. I thought about using GoSub,
but how do I get the caller back into VoiceMail?
I've done a bunch of searching for this, but haven't found any general
solutions. Is it possible to do what I'm trying to achieve, or is there
a better approach?
This is Asterisk 11.13.
Cheers,
Andrew
--
Andrew Ruthven, Wellington, New Zealand
MIITP, CITPNZ
At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Card : http://qr.catalyst.net.nz/907675e1
Cloud : NZs only real cloud - https://catalyst.net.nz/cloud
GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863
LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.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] Asterisk 13.10.0 Now Available
21.7.2016, 20:38, Asterisk Development Team kirjoitti: Bugs fixed in this release: --- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by Alexander Traud) Now it's possible to use dtls_cipher settings such like: dtls_cipher=ALL:!SSLv3 or dtls_cipher=HIGH:!SSLv3 Thank you! Best, Teijo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 13.10.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.10.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.10.0 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:
Improvements made in this release:
---
* ASTERISK-26088 - Investigate heavy memory utilization by
res_pjsip_pubsub (Reported by Richard Mudgett)
* ASTERISK-26011 - [patch]PJSIP: add "via_addr", "via_port",
"call_id" to contacts (Reported by Alexei Gradinari)
* ASTERISK-25994 - [patch]res_pjsip: module load priority
(Reported by Alexei Gradinari)
* ASTERISK-25931 - PJSIP: add "reg_server" to contacts. (Reported
by Alexei Gradinari)
* ASTERISK-25835 - Authentication using 'Username' field from
Digest (Reported by Ross Beer)
* ASTERISK-25930 - PJSIP: disable multi domain to improve realtime
performace (Reported by Alexei Gradinari)
Bugs fixed in this release:
---
* ASTERISK-26160 - pjsip: Updated->Reachable during qualify
(Reported by Matt Jordan)
* ASTERISK-26177 - func_odbc: Database handle is kept when it
should be released (Reported by Leandro Dardini)
* ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
due to server timeout (Reported by Ross Beer)
* ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to
v21_details (Reported by Corey Farrell)
* ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a
self-comparison (Reported by George Joseph)
* ASTERISK-26138 - chan_unistim: Under FreeBSD, chan_unistim
generates a compile error (Reported by George Joseph)
* ASTERISK-26128 - Alembic scripts are failing (Reported by Mark
Michelson)
* ASTERISK-26139 - test_res_pjsip_scheduler: Compile failure if
pjproject isn't installed in a system location (Reported by
George Joseph)
* ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
(Reported by Alexander Traud)
* ASTERISK-26127 - res_pjsip_session: Crash due to race condition
between res_pjsip_session unload and timer (Reported by Joshua
Colp)
* ASTERISK-26083 - ARI: Announcer channels staying around after
playback to a bridge is finished (Reported by Per Jensen)
* ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in
http.conf (Reported by Alexander Traud)
* ASTERISK-26069 - Asterisk truncates To: header, dropping the
closing '>' (Reported by Vasil Kolev)
* ASTERISK-26097 - [patch] CLI: show maximum file descriptors
(Reported by Alexander Traud)
* ASTERISK-25262 - Memory leak when a caller channel does multiple
dials and CEL is enabled (Reported by Etienne Lessard)
* ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 after
Remotely bridged channels (Reported by Niklas Larsson)
* ASTERISK-26096 - res_hep: Crash when configuration file is
missing (Reported by Niklas Larsson)
* ASTERISK-26089 - Invalid security events during boot using PJSIP
Realtime (Reported by Scott Griepentrog)
* ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by
Ross Beer)
* ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B.
Davis)
* ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
against libsrtp-1.5.0 (Reported by Patrick Laimbock)
* ASTERISK-26091 - [patch] ar cru creates warning, instead use ar
cr (Reported by Alexander Traud)
* ASTERISK-26070 - ari/channels: Creating a local channel without
an originator adds all audio formats to it's capabilities
(Reported by George Joseph)
* ASTERISK-26078 - core: Memory leak in logging (Reported by
Etienne Lessard)
* ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not ordered
properly (Reported by Ross Beer)
* ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work -
documentation needs clarification for when read/write is
possible (Reported by Private Name)
* ASTERISK-25777 - data race in threadpool (Reported by Badalian
Vyacheslav)
* ASTERISK-26038 - 'make install' doesn't seem to install OS/X
init files (Reported by Tzafrir Cohen)
* ASTERISK-26029 - parking: ast_parking_park_call should return
parking_space instead of parking_exten (Reported by Diederik de
Groot)
* ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
LAST_INSERT_ID() always returns zero. (Reported by Edwin
Vandamme)
* ASTERISK-25941 - chan_pjsip: Crash on an immediate SIP final
response (Reported by Javier Riveros )
* ASTERISK-26014 - res_sorcery_astdb: Make tolerant of unknown
fields (Reported by Joshua Colp)
* ASTERISK-24986 - keepalive INFO packages ignored by asterisk
(Reported by Ilya Trikoz)
* ASTERISK-26034 - T.38 passthrough problem behind fir
[asterisk-users] Asterisk 11.23.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.23.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.23.0 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-26141 - res_fax: fax_v21_session_new leaks reference to v21_details (Reported by Corey Farrell) * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported by George Joseph) * ASTERISK-26138 - chan_unistim: Under FreeBSD, chan_unistim generates a compile error (Reported by George Joseph) * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by Alexander Traud) * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported by Alexander Traud) * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '>' (Reported by Vasil Kolev) * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by Alexander Traud) * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 (Reported by Patrick Laimbock) * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported by Alexander Traud) * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files (Reported by Tzafrir Cohen) * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early nosignal packet (Reported by George Joseph) * ASTERISK-26030 - call cut because of double Session-Expires header in re-invite after proxy authentication is required (Reported by George Joseph) * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported by Tzafrir Cohen) * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is in the process of being recorded during reload (Reported by John Campbell) * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself (Reported by Jonathan R. Rose) * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName (Reported by Javier Acosta) * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive sometimes two calls (Reported by nik600) * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers table unless rt (Reported by Jaco Kroon) * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of app_queue.c (Reported by Sébastien Couture) * ASTERISK-25874 - app_voicemail: Stack buffer overflow in test_voicemail_notify_endl (Reported by Badalian Vyacheslav) * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding them to the local hangupcauses via ast_channel_hangupcause_hash_set (Reported by Jaco Kroon) * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton) Improvements made in this release: --- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad de Wet) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0 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] Is it possible to change the default format for ConfBridge recordings?
We have a customer who does significant ConfBridge recording every day. They are concerned about the size of the recording that will accumulate. >From the confbridge.conf.sample file, it mentions "the default format is 8khz >slinear" It is possible to change that "default format" and if so, how would I go about doing this? Have a great day! Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] 1.8.32.3 - billsec field does not increment after call answer - what triggers it?
Stefan Viljoen wrote: Only this one trunk consistenly has this problem for all calls received over it. The trunk provider is using sippy on their side. What setting / config option for the particular SIP "problem trunk" have my trunk provider changed on their side to stop Asterisk from recognising that a call has been answered when it comes in over that trunk? It appears some SIP traffic is not being sent by them (or not received by my Asterisk) that indicates to it a call has been ANSWERED and that it must start the billsec timer? I can't really speak for the provider but some numbers will stay in inband progress (unanswered) for a bit. Some toll-frees for example. The specific SIP message that would show it as answered would be a 200 OK to the INVITE we sent though. If you provided the SIP log then we could see. 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
