Hello list,

Kevin I agree with you on independent monitored entity for A leg while the
outbound leg has separate QoS measures. But after this thread I went to my
monitoring tool and saw that for some calls on the same asterisk setup I had
no RTP or RTCP while there were calls with both RTP and RTCP captured as
well.

Since I’ve a SIP proxy on top of asterisk servers layers, could it be
possible that RTP and RTCPs bypass asterisk (media redirect) and that’s why
I see RTCPs and RTPs logged into monitoring tool while those call who
couldn’t redirect/bypass media from asterisk don’t show any RTCPs!?

Sammy can you provide further details of your setup please!

Regards,
Gohar

Hi,
I’d like to implement some way of controlling remote SIP clients while
in a call, to execute remote commands.

The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is connected
to the previous one

I’m wondering if Asterisk is relaying DTMF (SIP info or RTP) from the
caller to the callees. I found option ‘F’ for MeetMe, but I have no
control on Page().

TIA,
Matteo

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following is a sample of the issues resolved in this release:

* AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
Reported by: Catalin Sanda

* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing
to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely. This causes a variety of negative side
effects, depending on when the loop exits. This patch handles the frame by
essentially swallowing the frame in the local loop, as the current channel
drivers expect the RTP bridge to handle the frame, and, in the case of the
local bridge loop, no additional action is necessary.
(closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
by: Matt Jordan

* Fix timing source dependency issues with MOH. Prior to this patch,
res_musiconhold existed at the same module priority level as the timing
sources that it depends on. This would cause a problem when music on
hold was reloaded, as the timing source could be changed after
res_musiconhold was processed. This patch adds a new module priority
level, AST_MODPRI_TIMING, that the various timing modules are now loaded
at. This now occurs before loading other resource modules, such
that the timing source is guaranteed to be set prior to resolving
the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,
Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patched by elguero

* Fix RTP reference leak. If a blind transfer were initiated using a
REFER without a prior reINVITE to place the call on hold, AND if Asterisk
were sending RTCP reports, then there was a reference leak for the
RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali

* Fix blind transfers from failing if an ‘h’ extension
is present. This prevents the ‘h’ extension from being run on the
transferee channel when it is transferred via a native transfer
mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported
by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
Mark Michelson (license 5049)

* Restore call progress code for analog ports. Extracting sig_analog
from chan_dahdi lost call progress detection functionality. Fix
analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller Patched by Richard Miller

* Fix regression that ‘rtp/rtcp set debup ip’ only works when a port
was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by:
Walter Doekes

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0

Thank you for your continued support of Asterisk!

Hello,

I am trying to deposit a voicemail message(using voicemail() application) for a subscriber using asterisk-1.8.7.1. But i am facing  aproblem in the rtp port allocation for a session due to which ’488 Not Acceptable’ response is sent towards the client end.  Following are error messages:

[Jan 18 12:43:59] ERROR[19164] res_rtp_asterisk.c: Failed to Allocate port 7660 for RTP instance ’0x1a75ab98′
[Jan 18 12:43:59] ERROR[19164] res_rtp_asterisk.c: Oh dear… we couldn’t allocate a port (x=7662)7660 for RTP instance ’0x1a75ab98′. errno 99
[Jan 18 12:43:59] DEBUG[19164] rtp_engine.c: Engine ‘asterisk’ failed to setup RTP instance ’0x1a75ab98′
[Jan 18 12:43:59] DEBUG[19164] rtp_engine.c: Destroyed RTP instance ’0x1a75ab98′
[Jan 18 12:43:59] DEBUG[19164] chan_sip.c: ERROR: failed to allocate rtp instance
[Jan 18 12:43:59] DEBUG[19164] chan_sip.c: Could not initialize RTP instance for dialog: 800E51A5-1140-E111-A216-001A4B4698C3@10.34.77.90

Please  find attached the log file  for more information.

Regards,
Shalu

messages_rtp_port_allocation_problem.zip

Hello,

I am trying to run load on asterisk server(version 1.8.7.1) through SIPp tool for the voicemail() application. But I am facing a lot of problems. I tried running 1000 calls from SIPp for 100 subscribers (10 messages for each subscriber). I am using odbc storage for the messages.

Following warnings/errors are coming on the asterisk server:

Jan 11 11:30:49] WARNING[22924] app.c: Failed to lock path ‘/var/spool/asterisk/voicemail/default/27/INBOX’: File exists
[Jan 11 11:30:49] ERROR[22924] app_voicemail.c: Couldn’t lock directory /var/spool/asterisk/voicemail/default/27/INBOX.  Voicemail will be lost.

Sometimes I have seen that  .lock file remains in the INBOX folder for a particular subscriber. I wanted to know why this .lock file is not deleted. Is this a bug or I am missing something in the configuration.

[ Jan 11 11:30:50] WARNING[22874] app_voicemail.c: Open of sound file ‘/var/spool/asterisk/voicemail/default/2/INBOX/msg0011.gsm’ failed: No such file or directory 

[Jan 11 11:30:50] WARNING[ 23109] app .c: No audio available on SIP/172.16.129.13:9027-00000206??

[ Jan 11 11:30:17] ERROR[22122] res_rtp_asterisk.c: Oh dear… we couldn’t allocate a port (x=6460)6460 for RTP instance ’0x2aaacd6454c8′. errno 98

I have given the rtp port range as 6000 to 8000 in rtp.conf. Is this not sufficient for running 1000 calls.

The SIPp command which I am running  is as follows:

/sipp -m 10 -r 1 -rp 1000 -d 45 -mp 7000 -p 9000 -bg -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 –trace_err
usleep 80000
./sipp -m 10 -r 1 -rp 1000 -d 45 -mp 7004 -p 9001 -bg -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 2 172.16.129.14 –trace_err
usleep 80000
./sipp -m 10 -r 1 -rp 1000 -d 45 -mp 7008 -p 9002 -bg -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 3 172.16.129.14 –trace_err
usleep 80000
./sipp -m 10 -r 1 -rp 1000 -d 45 -mp 7012 -p 9003 -bg -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 4 172.16.129.14 –trace_err