Re: [asterisk-users] Need more meetme users -- hitting some limit

[Steve Edwards]

On Mon, 31 Mar 2014, Shaun Ruffell wrote:


If you're looking to reduce the CPU overhead of processing meetme
conferences, this email from awhile ago may be of some help:

http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/51750/focus=51777


Thanks for the clue. I can hit my target of 512 on an Intel E3-1240v3 with 
'pre-packaged' Asterisk so I'm good for now.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Unable to build DAHDI-Linux in mock chroot

[Anthony Messina]

On Sunday, March 30, 2014 02:24:35 PM Anthony Messina wrote:
> On Sunday, March 30, 2014 07:07:47 PM Tzafrir Cohen wrote:
> > On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote:
> > > On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote:
> > > > Unfortunately, after
> > > >
> > > > 
> > > >
> > > > http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc
> > > > 1c
> > > > 1fb1 2cc0661f3810ef47ad33206b2e398
> > > >
> > > > 
> > > >
> > > > I am unable to build DAHDI-Linux in a mock chroot for packaging
> > > > purposes.  I  believe this is related to the Makefile calling
> > > > install_firmware with only 2 args, where install_firmware is a shell
> > > > script
> > > > with DESTDIR set to $3, which is empty.
> > > >
> > > > 
> > > >
> > > > In this case, the DESTDIR evaluates to "/usr/lib/hotplug/firmware",
> > > > rather 
> > > > than "/usr/lib/hotplug/firmware".
> > > >
> > > > 
> > > > 
> > > >
> > > > make -C drivers/dahdi/firmware hotplug-install 
> > > > DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 
> > > > HOTPLUG_FIRMWARE=yes
> > > > make[1]: Entering directory `/builddir/build/BUILD/dahdi-
> > > > linux-2.9.1/drivers/dahdi/firmware'
> > > > mkdir -p /builddir/build/BUILDROOT/dahdi-
> > > > linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware
> > > > mkdir -p /builddir/build/BUILDROOT/dahdi-
> > > > linux-2.9.1-1.fc20.x86_64/lib/firmware
> > > > Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories
> > > > install: cannot create regular file '/usr/lib/hotplug/firmware': No
> > > > such
> > > > file  or directory
> > > > make[1]: *** [hotplug-install] Error 1
> > > > make[1]: Leaving directory `/builddir/build/BUILD/dahdi-
> > > > linux-2.9.1/drivers/dahdi/firmware'
> > > > make: *** [install-firmware] Error 2
> > >
> > > 
> > >
> > > https://issues.asterisk.org/jira/browse/DAHLIN-337
> >
> > 
> >
> > Thanks for your report. I hope to get it fixed soon.
> > I should note that this specific target does not belong in a proper
> > chroot build, as it downloads from outside. How can I get those firmware
> > files properly included?
> 
> This is the spec file I use:
> https://messinet.com/rpms/browser/dahdi-linux/dahdi-linux.spec

DAHDI-Linux-2.9.1.1 fixes this issue. Thank you.  -A

-- 
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] Need more meetme users -- hitting some limit

[Shaun Ruffell]

On Fri, Mar 21, 2014 at 11:26:22AM -0700, Steve Edwards wrote:
> On Fri, 21 Mar 2014, Steve Totaro wrote:
> 
> >I found below here:
> > http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
> >
> >If you have too many conferences, one CPU may not be able to mix all the
> >audio and you will have audio problems even if there are 7+ other CPUs
> >that are essentially idle while waiting for one CPU to mix everything. You
> >should be able to handle 512 conference participants on a modern server
> >system without problem. The current trunk of DAHDI linux limits the number
> >of open pseudo channels to 512 for this reason. [1]
> 
> With 312 calls distributed across 100 meetmes, 'top' shows 1 core at 32%, 1
> core at 6% and the rest basically idle.
> 
> So it looks like meetme() is still a single CPU application, but I have
> plenty of CPU headroom.
> 
> Coincidentally, 512 is my target. Any clues on how to get 200 more?

Steve,

If you're looking to reduce the CPU overhead of processing meetme
conferences, this email from awhile ago may be of some help:
 
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/51750/focus=51777

Cheers,
Shaun

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] DAHDI-Linux v2.9.1.1 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.9.1.1
dahdi-linux-complete-2.9.1.1+2.9.1

This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

Fix for bug preventing DAHDI from building in a chroot environment

Issues closed in this release:
DAHLIN-337

Shortlog of dahdi-linux changes since v2.9.1:
Tzafrir Cohen (1):
  firmware: Honor DESTDIR when installing firmware.

The diffstat from the dahdi-linux v2.9.1 release:
 drivers/dahdi/firmware/Makefile | 31 ---
 1 file changed, 16 insertions(+), 15 deletions(-)

For a full list of changes in these releases, please see the shortlog at:
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.9.1.1

Issues found in this release can be reported in the DAHDI-Linux [1] and
DAHDI-Tools [2] projects at https://issues.asterisk.org/jira

[1] https://issues.asterisk.org/jira/browse/DAHLIN
[2] https://issues.asterisk.org/jira/browse/DAHTOOL

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] CLI command to see if SRTP is active?

[Rusty Newton]

On Mon, Mar 31, 2014 at 1:26 PM, Rusty Newton  wrote:
> On Fri, Mar 28, 2014 at 7:39 PM, Patrick Laimbock  
> wrote:
>> Hi,
>>
>> I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI
>> command to see if SRTP is active on a channel/call. I went through sip show
>> ... and core show channel... and did not see any mentioning of SRTP while
>> there is an SRTP call active.
>
> I don't have any encrypted calls up in front of me at this second to
> provide an example, however if you are just wanting to verify SRTP is
> active for a call you will see SRTP related messages on the CLI if you
> turn up DEBUG message verbosity and have it going to the console, or
> else possibly with output from "rtp set debug on".
>
> As for the "show channels" type commands, it may say something about
> encryption rather than "SRTP" directly. I'll take a look later if I
> get a chance.

I see your issue on the tracker now, posting the link here for the
sake of those who read the archives
https://issues.asterisk.org/jira/browse/ASTERISK-23564


-- 
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] CLI command to see if SRTP is active?

[Rusty Newton]

On Fri, Mar 28, 2014 at 7:39 PM, Patrick Laimbock  wrote:
> Hi,
>
> I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI
> command to see if SRTP is active on a channel/call. I went through sip show
> ... and core show channel... and did not see any mentioning of SRTP while
> there is an SRTP call active.

I don't have any encrypted calls up in front of me at this second to
provide an example, however if you are just wanting to verify SRTP is
active for a call you will see SRTP related messages on the CLI if you
turn up DEBUG message verbosity and have it going to the console, or
else possibly with output from "rtp set debug on".

As for the "show channels" type commands, it may say something about
encryption rather than "SRTP" directly. I'll take a look later if I
get a chance.

-- 
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Function REGEX

[Rafael dos Santos Saraiva]

All working fine.
Thank you for your help.


Att,
*Rafael dos Santos Saraiva*



2014-03-31 12:29 GMT-03:00 Eric Wieling :

> Here is an example from one of my production dialplans
>
> same =>
> n,ExecIf(${REGEX("^1205|^1256|^1850|^1718|^1212|^1917|^1347|^1646|^1929"
> ${CALLERID(num)})}]?Hangup)
>
> Assuming you meant 0-9 and not the literal X (which means nothing special
> in regular expressions):
>
> same => n,ExecIf(${REGEX("^[0-9]5123$" ${EXTEN})}]?Hangup)
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos
> Saraiva
> Sent: Monday, March 31, 2014 11:23 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Function REGEX
>
> Hi
>
> I need help to use the function REGEX. My question is if is possible test
> a expression as [X123 == 5123] ( If an extension corresponding to a
> previously defined regular expression). I saw various examples about this
> function, but nothing as the my  needs. I do not understanding exactly how
> to works this function.
>
> Thank's
>
> Att,
> Rafael dos Santos Saraiva
>  
>
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Re: [asterisk-users] Function REGEX

[Eric Wieling]

Here is an example from one of my production dialplans

same => 
n,ExecIf(${REGEX("^1205|^1256|^1850|^1718|^1212|^1917|^1347|^1646|^1929" 
${CALLERID(num)})}]?Hangup)

Assuming you meant 0-9 and not the literal X (which means nothing special in 
regular expressions):

same => n,ExecIf(${REGEX("^[0-9]5123$" ${EXTEN})}]?Hangup)


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos 
Saraiva
Sent: Monday, March 31, 2014 11:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Function REGEX

Hi

I need help to use the function REGEX. My question is if is possible test a 
expression as [X123 == 5123] ( If an extension corresponding to a previously 
defined regular expression). I saw various examples about this function, but 
nothing as the my  needs. I do not understanding exactly how to works this 
function.

Thank's

Att,
Rafael dos Santos Saraiva
  

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[asterisk-users] Function REGEX

[Rafael dos Santos Saraiva]

Hi

I need help to use the function REGEX. My question is if is possible test a
expression as [X123 == 5123] ( If an extension corresponding to a
previously defined regular expression). I saw various examples about this
function, but nothing as the my  needs. I do not understanding exactly how
to works this function.

Thank's

Att,
*Rafael dos Santos Saraiva*

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Re: [asterisk-users] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways

[Joshua Colp]

Matt Rabbitt wrote:

What would need to be changed in the source code to accommodate this?
  Can the imageattr attribute be hard coded into
h264_format_attr_sdp_generate() in res_format_attr_h264.c?


A lot. Yes, you could hard code it.

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Re: [asterisk-users] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways

[Matt Rabbitt]

What would need to be changed in the source code to accommodate this?  Can
the imageattr attribute be hard coded into h264_format_attr_sdp_generate()
in res_format_attr_h264.c?


On Mon, Mar 31, 2014 at 9:07 AM, Joshua Colp  wrote:

> Matt Rabbitt wrote:
>
>> We are experiencing an issue with our Cisco 9971 and 8945 phones where
>> H264 video calls are connecting at 176x144 resolution instead of
>> 640x480.  Soft clients can connect at higher resolutions and the 9971
>> can even receive video at a higher resolution (although it still sends
>> 176x144).
>>
>> I contacted one of the developers and he suggested the passthrough of
>> SDP attributes is not working correctly.  Has anyone else experienced
>> this problem?  We're running Asterisk 11.8.1.
>>
>> Below are the video parts of the sip debug for one of the phones during
>> a video call.  Should I be seeing the "a=imageattr" in the SIP OK message?
>>
>
> It looks as though the passthrough for "fmtp" is indeed working but as the
> "imageattr" attribute is currently unsupported/not used/not passed through
> it is probably causing your resolution problem.
>
> 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
>
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Re: [asterisk-users] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways

[Joshua Colp]

Matt Rabbitt wrote:

We are experiencing an issue with our Cisco 9971 and 8945 phones where
H264 video calls are connecting at 176x144 resolution instead of
640x480.  Soft clients can connect at higher resolutions and the 9971
can even receive video at a higher resolution (although it still sends
176x144).

I contacted one of the developers and he suggested the passthrough of
SDP attributes is not working correctly.  Has anyone else experienced
this problem?  We're running Asterisk 11.8.1.

Below are the video parts of the sip debug for one of the phones during
a video call.  Should I be seeing the "a=imageattr" in the SIP OK message?


It looks as though the passthrough for "fmtp" is indeed working but as 
the "imageattr" attribute is currently unsupported/not used/not passed 
through it is probably causing your resolution problem.


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

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[asterisk-users] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways

[Matt Rabbitt]

We are experiencing an issue with our Cisco 9971 and 8945 phones where H264
video calls are connecting at 176x144 resolution instead of 640x480.  Soft
clients can connect at higher resolutions and the 9971 can even receive
video at a higher resolution (although it still sends 176x144).

I contacted one of the developers and he suggested the passthrough of SDP
attributes is not working correctly.  Has anyone else experienced this
problem?  We're running Asterisk 11.8.1.

Below are the video parts of the sip debug for one of the phones during a
video call.  Should I be seeing the "a=imageattr" in the SIP OK message?



<--- SIP read from UDP:10.168.154.71:5060 --->
INVITE sip:7872@10.162.26.15;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.168.154.71:5060;branch=z9hG4bK1182b2d3
From: "Shawn Hughes" ;tag=20bbc0df35ef052672e68696-0b174da0
To: 
Call-ID: 20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71
Max-Forwards: 70
Date: Fri, 28 Mar 2014 13:51:41 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP8945/9.4.1
Contact: ;video
Authorization: Digest username="7871",realm="asterisk",uri="
sip:7872@10.162.26.15
;user=phone",response="f51a7522b01c90b81509d2274e9b69bb",nonce="5b43e5a6",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow:
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported:
replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 685
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 27778 0 IN IP4 10.168.154.71
s=SIP Call
t=0 0
m=audio 10032 RTP/AVP 0 8 18 102 9 116 101
c=IN IP4 10.168.154.71
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 10034 RTP/AVP 97
c=IN IP4 10.168.154.71
b=TIAS:200
a=rtpmap:97 H264/9
a=fmtp:97
profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200
a=imageattr:97 send [x=640,y=480] [x=640,y=360] [x=352,y=288] [x=176,y=144]
recv [x=640,y=480]
a=sendrecv
<->
--- (19 headers 24 lines) ---
Sending to 10.168.154.71:5060 (no NAT)
Using INVITE request as basis request -
20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71
Found peer '7871' for '7871' from 10.168.154.71:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 102
Found RTP audio format 9
Found RTP audio format 116
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format L16 for ID 102
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Found RTP video format 97
Found video description format H264 for ID 97
Capabilities: us - (gsm|ulaw|alaw|g722|h264), peer -
audio=(ulaw|alaw|g729|ilbc|g722|slin16)/video=(h264)/text=(nothing),
combined - (ulaw|alaw|g722|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.168.154.71:10032
Peer video RTP is at port 10.168.154.71:10034
Looking for 7872 in from-internal (domain 10.162.26.15)
list_route: hop: 



SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.154.71:5060
;branch=z9hG4bK1182b2d3;received=10.168.154.71
From: "Shawn Hughes" ;tag=20bbc0df35ef052672e68696-0b174da0
To: ;tag=as1c2f9ae5
Call-ID: 20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.8.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: 
Content-Type: application/sdp
Content-Length: 467

v=0
o=root 283568327 283568327 IN IP4 10.162.26.15
s=Asterisk PBX 11.8.1
c=IN IP4 10.162.26.15
b=CT:3600
t=0 0
m=audio 13434 RTP/AVP 0 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 15496 RTP/AVP 97
a=rtpmap:97 H264/9
a=fmtp:97
profile-level-id=428014;max-mbps=36000;max-fs=1200;packetization-mode=0;level-asymmetry-allowed=1
a=sendrecv
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Re: [asterisk-users] IAXModem or T38Modem?

[James Cloos]

> "j" == joakimsen   writes:

j> I wouldn't mind if someone posted on the list a known working provider
j> with the proper configuration to use T.38. In my case I don't consider
j> it an issue with the provider because they sent the proper T.38
j> Invite, but Asterisk IMO does not know how to handle it.

Are you using a single credential-tuple with the provider?

If the provider supports T.38 and if you can separate out fax lines,
there is no need to stick asterisk between them and t38modem.  Just have
t38modem access the provider directly.  Hylafax will handle the rest.

(Look for things like sub-account, peer and/or trunk configs.)

-JimC
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James Cloos  OpenPGP: 1024D/ED7DAEA6

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Re: [asterisk-users] Duplicate incoming channel into two outgoing channels

[Klaus Darilion]

On 27.03.2014 10:39, jg wrote:

Wouldn't it make more sense to handle this by just monitoring the calls
and doing everything else with normal data processing?


Basically yes, but the whole idea is a workaround to fix issues in 
legacy systems.


klaus

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