[asterisk-users] get CDR log after disposition status answer
Hi, Im planning to store CDR log to another database, so I used these Dialplan code first: exten = _X.,n,DeadAGI(xml-rpc_sendCDRLog.php) or exten = h,n,AGI(xml-rpc_sendCDRLog.php) Actually it seems both of them work, But not in ANSWER disposition status.(when call answred, AGI not run) Please let me know how can I run my AGI code when user answer the call and then hangup it. -- _ -- 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] ConfBridge performance problem...?
Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants that they've seen in a conference. I was never able to get more than 8 participants (mixed G722 and G711a) on a conference (actually that's per server limit) with almost all settings on default, except for dsp_drop_silence and denoise which are enabled. I tested on Debian squeeze, 64-bit, quad-core Xeon server @2.4GHz and also on another virtual server with similar processor (just one core available to the VM). While this is not the latest and greatest CPU, I would certainly expect it to handle more than 8 calls. To be honest, I was in fact able to get it working for up to 20 participants (most with G711), when I switched from res_timing_timerfd to res_timing_dahdi and turned off denoise, but that's still not normal I believe, especially with most participants on mute and with dps_drop_silence enabled and nothing else running on the server. The problem itself is, that once I get over the critical number of participants, the voice starts to break up and it's impossible to understand the person who's talking. This is certainly not bandwidth related because all tests were made on the LAN and besides I could see that the CPU was sometime close to 100%. Did someone observe something similar? BTW, once the first participant enters the conference I start seeing probably over 50 messages per second saying: bridging.c:757 bridge_channel_join_multithreaded: Going into a multithreaded waitfor for bridge channel 0x292d708 of bridge 0x28f3658 Best, Hristo -- _ -- 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] ConfBridge performance problem...?
Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command dahdi_test? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants that they've seen in a conference. I was never able to get more than 8 participants (mixed G722 and G711a) on a conference (actually that's per server limit) with almost all settings on default, except for dsp_drop_silence and denoise which are enabled. I tested on Debian squeeze, 64-bit, quad-core Xeon server @2.4GHz and also on another virtual server with similar processor (just one core available to the VM). While this is not the latest and greatest CPU, I would certainly expect it to handle more than 8 calls. To be honest, I was in fact able to get it working for up to 20 participants (most with G711), when I switched from res_timing_timerfd to res_timing_dahdi and turned off denoise, but that's still not normal I believe, especially with most participants on mute and with dps_drop_silence enabled and nothing else running on the server. The problem itself is, that once I get over the critical number of participants, the voice starts to break up and it's impossible to understand the person who's talking. This is certainly not bandwidth related because all tests were made on the LAN and besides I could see that the CPU was sometime close to 100%. Did someone observe something similar? BTW, once the first participant enters the conference I start seeing probably over 50 messages per second saying: bridging.c:757 bridge_channel_join_multithreaded: Going into a multithreaded waitfor for bridge channel 0x292d708 of bridge 0x28f3658 Best, -- _ -- 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] ConfBridge performance problem...?
Sorry - I just read you alsways checked the cpu usage. Are all cores at 100%? Is it the atserisk process which consumes it all? Am 06.02.2013 13:54, schrieb Thorsten Göllner: Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command dahdi_test? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants that they've seen in a conference. I was never able to get more than 8 participants (mixed G722 and G711a) on a conference (actually that's per server limit) with almost all settings on default, except for dsp_drop_silence and denoise which are enabled. I tested on Debian squeeze, 64-bit, quad-core Xeon server @2.4GHz and also on another virtual server with similar processor (just one core available to the VM). While this is not the latest and greatest CPU, I would certainly expect it to handle more than 8 calls. To be honest, I was in fact able to get it working for up to 20 participants (most with G711), when I switched from res_timing_timerfd to res_timing_dahdi and turned off denoise, but that's still not normal I believe, especially with most participants on mute and with dps_drop_silence enabled and nothing else running on the server. The problem itself is, that once I get over the critical number of participants, the voice starts to break up and it's impossible to understand the person who's talking. This is certainly not bandwidth related because all tests were made on the LAN and besides I could see that the CPU was sometime close to 100%. Did someone observe something similar? BTW, once the first participant enters the conference I start seeing probably over 50 messages per second saying: bridging.c:757 bridge_channel_join_multithreaded: Going into a multithreaded waitfor for bridge channel 0x292d708 of bridge 0x28f3658 -- _ -- 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] ConfBridge performance problem...?
Did you check asterisk -rx core show translation recalc 10 Am 06.02.2013 13:56, schrieb Thorsten Göllner: Sorry - I just read you alsways checked the cpu usage. Are all cores at 100%? Is it the atserisk process which consumes it all? Am 06.02.2013 13:54, schrieb Thorsten Göllner: Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command dahdi_test? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants that they've seen in a conference. I was never able to get more than 8 participants (mixed G722 and G711a) on a conference (actually that's per server limit) with almost all settings on default, except for dsp_drop_silence and denoise which are enabled. I tested on Debian squeeze, 64-bit, quad-core Xeon server @2.4GHz and also on another virtual server with similar processor (just one core available to the VM). While this is not the latest and greatest CPU, I would certainly expect it to handle more than 8 calls. To be honest, I was in fact able to get it working for up to 20 participants (most with G711), when I switched from res_timing_timerfd to res_timing_dahdi and turned off denoise, but that's still not normal I believe, especially with most participants on mute and with dps_drop_silence enabled and nothing else running on the server. The problem itself is, that once I get over the critical number of participants, the voice starts to break up and it's impossible to understand the person who's talking. This is certainly not bandwidth related because all tests were made on the LAN and besides I could see that the CPU was sometime close to 100%. Did someone observe something similar? BTW, once the first participant enters the conference I start seeing probably over 50 messages per second saying: bridging.c:757 bridge_channel_join_multithreaded: Going into a multithreaded waitfor for bridge channel 0x292d708 of bridge 0x28f3658 -- _ -- 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] Set(CALLERID(num-pres)=prohib) / AGI version
Hi, on this site http://www.voip-info.org/wiki/view/Asterisk+func+callerid you can read, that since Atserisk 1.8 the command (in dialplan) to hide the caller id is: Set(CALLERID(num-pres)=prohib) I tried to implement it into my AGI-Script, but with no success. Can please anyone give me a hint, what is wrong with it: Set CALLERID(num-pres) prohib or Set CALLERID(num-pres)=prohib Both commands lead into: 510 Invalid or unknown command Besr regards -Thorsten- -- _ -- 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] RPM updates
On 28 Jan 2013, at 13:55, Steven Howes wrote: Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers for the replies regarding alternative repos. I'm looking to keep using the Digium ones, but they're still broken. Guess I'll just have to wait until someone at Digium notices :S Steve -- _ -- 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] Set(CALLERID(num-pres)=prohib) / AGI version
On Wed, 6 Feb 2013, Thorsten Göllner wrote:
I tried to implement it into my AGI-Script, but with no success. Can please
anyone give me a hint, what is wrong with it:
Set CALLERID(num-pres) prohib
or
Set CALLERID(num-pres)=prohib
Both commands lead into:
510 Invalid or unknown command
I'm just a 1.2 Luddite, but...
Who's library/framework are you using?
Neither of the commands you show above are valid AGI commands.
Curiously, I've never tried to set caller ID (or its options) in an AGI,
I've only set channel variables that ended up setting CID in the dialplan.
If you were reading the variables, the command would look like:
'get full variable ${CALLERID(num-pres)}'
Maybe you could try something like:
'set variable CALLERID(num-pres) prohib'
(I don't see a 'set full variable' AGI command.)
How about a console log with verbose and debug cranked up and with AGI
debug enabled?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
_
-- 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] dahdi-channels.conf parameters
On Tue, Feb 05, 2013 at 02:09:12PM -0600, Hose wrote:
Hi,
I've always used dahdi-genconf to just create the dahdi-channels.conf
and since our PRI is fairly simple (just dump all the channels into one
group) it works with dialing with dahdi/g1/(number). I'm trying to
understand the file though for my own reference.
It seems the file looks like this:
group=0,11
context=from-pstn
switchtype = national
signalling = pri_cpe
channel = 1-23
context = default
group = 63
So what I don't get is why group is specified with 0,11. Is that groups
0 and groups 11? And then it has a random group = 63 at the end. And I
dial with group 1 (dahdi/g1), but it seems to work? :) It's completely
confused me as to why this actually works.
'group = 63' is a work around an old (and long-ago fixed, I believe) in
chan_dahdi (or maybe chan_zap - and never made it to chan_dahdi?). You
can just write 'group = ' instead to reset the groups.
It would be even nicer to write the above as:
[dahdi-span-1]
group=0,11
context=from-pstn
switchtype = national
signalling = pri_cpe
dahdichan = 1-23
('dahdi-span-1' is an arbitrary title). This only works for Asterisk
= 1.6.0, and thus I never got to switching dahdi_genconf to use it.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
--
_
-- 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] Set(CALLERID(num-pres)=prohib) / AGI version
Am 06.02.2013 16:02, schrieb Steve Edwards:
On Wed, 6 Feb 2013, Thorsten Göllner wrote:
I tried to implement it into my AGI-Script, but with no success. Can
please anyone give me a hint, what is wrong with it:
Set CALLERID(num-pres) prohib
or
Set CALLERID(num-pres)=prohib
Both commands lead into:
510 Invalid or unknown command
I'm just a 1.2 Luddite, but...
Who's library/framework are you using?
Neither of the commands you show above are valid AGI commands.
Curiously, I've never tried to set caller ID (or its options) in an
AGI, I've only set channel variables that ended up setting CID in the
dialplan.
If you were reading the variables, the command would look like:
'get full variable ${CALLERID(num-pres)}'
Maybe you could try something like:
'set variable CALLERID(num-pres) prohib'
(I don't see a 'set full variable' AGI command.)
How about a console log with verbose and debug cranked up and with AGI
debug enabled?
Thanks. But I found the right syntax now:
Exec Set CALLERID(num-pres)=prohib
This AGI-Command leads into 200 OK and I can verify, that outgoing
calls (SIP and DAHDI) are anonymous.
-Thorsten-
--
_
-- 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] problem
Hi every body; I want to intall some softwars working with my Asterisk server and I get these erreurs : * error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. /usr/sbin/grub-probe: error: cannot seek `/dev/sda'. dpkg: error processing grub-pc (--configure): subprocess installed post-installation script returned error exit status 1 * Please can any one let me know how can I resolve this problem; -- * **Élève Ingénieur INE2 à l'Institut National des Postes et Télécommunications * *INPT - Rabat - Maro*c * * * * *Responsable de la cellule Asterisk au **Club Electronique et Systemes Embarqués de l'INPT* *Membre du projet ilearn, SIFE INPT* * * * Tel : +212642398782 * -- _ -- 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] problem
Hi Are you sure that your hard drive sda, is ok? Looks like your hard drive is broken. On Wed, Feb 6, 2013 at 10:30 AM, brahim abidar abidarbah...@gmail.comwrote: Hi every body; I want to intall some softwars working with my Asterisk server and I get these erreurs : * error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. /usr/sbin/grub-probe: error: cannot seek `/dev/sda'. dpkg: error processing grub-pc (--configure): subprocess installed post-installation script returned error exit status 1 * Please can any one let me know how can I resolve this problem; -- * **Élève Ingénieur INE2 à l'Institut National des Postes et Télécommunications * *INPT - Rabat - Maro*c * * * * *Responsable de la cellule Asterisk au **Club Electronique et Systemes Embarqués de l'INPT* *Membre du projet ilearn, SIFE INPT* * * * Tel : +212642398782 * -- _ -- 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] Set(CALLERID(num-pres)=prohib) / AGI version
In article 51127171.2030...@ovm-group.com,
Thorsten Göllner t...@ovm-group.com wrote:
Am 06.02.2013 16:02, schrieb Steve Edwards:
On Wed, 6 Feb 2013, Thorsten Göllner wrote:
I tried to implement it into my AGI-Script, but with no success. Can
please anyone give me a hint, what is wrong with it:
Set CALLERID(num-pres) prohib
or
Set CALLERID(num-pres)=prohib
Both commands lead into:
510 Invalid or unknown command
I'm just a 1.2 Luddite, but...
Who's library/framework are you using?
Neither of the commands you show above are valid AGI commands.
Curiously, I've never tried to set caller ID (or its options) in an
AGI, I've only set channel variables that ended up setting CID in the
dialplan.
If you were reading the variables, the command would look like:
'get full variable ${CALLERID(num-pres)}'
Maybe you could try something like:
'set variable CALLERID(num-pres) prohib'
(I don't see a 'set full variable' AGI command.)
How about a console log with verbose and debug cranked up and with AGI
debug enabled?
Thanks. But I found the right syntax now:
Exec Set CALLERID(num-pres)=prohib
This AGI-Command leads into 200 OK and I can verify, that outgoing
calls (SIP and DAHDI) are anonymous.
Yes, that will work, but it is executing the Set() dialplan application.
AGI has a built-in command to set a variable, and it will be more efficient
to use that. I can confirm the following AGI syntax will work, as I have
used it to set other function-based values:
SET VARIABLE CALLERID(num-pres) prohib
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.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] problem
On Wed, Feb 6, 2013 at 9:30 AM, brahim abidar abidarbah...@gmail.comwrote: * error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. /usr/sbin/grub-probe: error: cannot seek `/dev/sda'. * I hope your hard drive is not /dev/sda. Although this could also be a configuration issue. * dpkg: error processing grub-pc (--configure): subprocess installed post-installation script returned error exit status 1 * And this indicates you're trying to install grub. If you're installing grub, you're probably starting from scratch on a new disk. You should look up a tutorial on how to test your hard drive, but an abbreviated tutorial is to get a Live DVD distro, burn it, boot off of it, and before mounting any partitions on /dev/sda, run badblocks -nvs /dev/sda on it. Let that run for however long it needs to; this will test all of the sectors on your drive to ensure they're working correctly. If it outputs any errors or starts writing out lots of numbers, you have a disk problem. Now, on to Asterisk: If your box is blank, you should just install a distribution designed for Asterisk: http://www.freepbx.org/freepbx-distro http://www.elastix.org/index.php/en/downloads/main-distro.html If this is an existing system, it sounds like you have a serious configuration issue or hardware problem, so you should head to the forums for whatever distro you're using (looks like some Debian variant). -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] problem
On Wednesday 06 February 2013, brahim abidar wrote: Hi every body; I want to intall some softwars working with my Asterisk server and I get these erreurs : * error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. /usr/sbin/grub-probe: error: cannot seek `/dev/sda'. dpkg: error processing grub-pc (--configure): subprocess installed post-installation script returned error exit status 1 * Please can any one let me know how can I resolve this problem; This isn't an Asterisk problem per se. It *could* be the beginning of a very nasty hardware problem (hard disk failure imminent) or it could be no more than a simple misconfiguration. It looks, from the messages, as though you are running Ubuntu or Debian. Is that so? Which release? And what output do you get for # mount and # fdisk -l ? -- AJS Answers come *after* questions. -- _ -- 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] encryption option when using realtime
Hi Everyone, I tried to load the extension configuration(extensions.conf) from mysql, but there is not the encrytion option in the configuration table of mysql. In a word when I enable the realtime, I can not set the encryption = yes, How can I do it? Any idear? Thanks in advanced. PS. in the directory /usr/src/asterisk-1.8.18.1/contrib/realtime/mysql sippeers.sql There isnot encryption field, if I add it by hand, it can not take effect. Regards. Hugo -- _ -- 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] Problem using ast_tls_cert script
Hi List, I'm trying to set my Asterisk 1.8.20.1 with TLS on CentOS 5.9, it was easy and straightforward with Debian 6.0.6, but when I introduce this command on CentOS: #./ast_tls_cert -C 10.200.108.17 -O MyCompany -d /etc/asterisk/keys/ I got this error message: hostname: Unknown host Same result happens when using server's hostname: #./ast_tls_cert -C ast-centos -O MyCompany -d /etc/asterisk/keys/ Where 'ast-centos' is the result of 'uname -n' I've followed instructions from: http://goalbound.blogspot.com/2012/05/configure-asterisk-18110-on-centos-55.html and https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Any hint would be appreciated! Elder D. Arohuanca DCAP Lima - Peru -- _ -- 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] Somewhat OT: Specific SIP packets can cause ethernet controller reset
While not strictly Asterisk related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner -- _ -- 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] Difference of outbound SDP offer behavior between 1.8.X and 11.X ?
Hi all,
I'm new to this mailing-list (and new to Asterisk in general) so please
don't blame me if this has already been posted here in the past (I tried
to check the archives but hey, they're quite big).
While testing a specific environment involving 2 Asterisks hosts, I
found different behaviors when using an Asterisk 1.8.X or an 11.2.1
(same on 11.1.2).
Here is the SIP case:
Friend1 --- Asterisk OutboundTrunk1
Everywhere, codecs allowed are: g722, g729, g711a, g711u.
Friend1 calls a pstn number for instance. It offers in the SDP included
with the INVITE g722/g711a/g711u (9 8 0).
The extensions.conf is configured as following:
exten = _XX,1,Dial(SIP/${EXTEN}@OutboundTrunk1)
The outbound SDP will be (these are real tcpdump captures):
A) Asterisk 1.8.15 cert1:
[...]
CSeq: 102 INVITE
User-Agent: SewanAstetrunk1
Date: Tue, 05 Feb 2013 14:16:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 322
v=0
s=Asterisk PBX 1.8.15-cert1
c=IN IP4 X.X.X.X
t=0 0
m=audio 33728 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
B) Asterisk 11.1.2:
[...]
CSeq: 102 INVITE
User-Agent: SewanAstetrunk1
Date: Tue, 05 Feb 2013 13:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 363
v=0
s=Asterisk PBX 11.1.2
c=IN IP4 X.X.X.X
t=0 0
m=audio 39102 RTP/AVP 9 18 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Imagine the remote trunk in this case isn't G722 capable, then
negotiates G729 with the Asterisk... Transcoding will be forced to G711a!
I reckon this difference in the offer is expected. May this be a bugfix
for Asterisk newer version?
Does anybody confirm this is not a bug?
I mean, in some terms, I really prefer the behavior of old Asterisk
1.8.15 because it will avoid transcoding in many many cases (Asterisk
11.X offers 18/g729 but he already knows Friend1 can't do g729 at the
time of initiating the outbound call). Moreover (and quite important),
it will avoid unnecessary usage of paid g729 transcoding in many cases
(meaning, cost savings).
Is there a way to make Asterisk 11.X behaves as 1.8.15? I.e. can we
restrict the SDP offer of the outbound call of at least what inbound leg
offered?
I tried to play with extensions.conf and SIP_OUTBOUND_CODEC (and
INBOUND), without luck.
Has anybody else wanted (I guess so) or even achieved this use case?
(And if yes, how?)
I have another question as well (which is related in a way):
is it possible to achieve this:
Friend1 invites Asterisk (offers for instance 9 8 0)
Asterisk invites OutboundTrunk1 (SDP: 8 0)
Asterisk got reply from OutboundTrunk1 (SDP: 8)
Asterisk then replies 200 OK with SDP 8 to Friend1 (and not 9, even if
capable, to avoid transcoding of course)
= Even if Asterisk behaves like a B2B should, is it possible to reply
the inbound considering the reply of the outbound leg?
I really think this should be achievable because Asterisk waits for the
outbound 200 to reply 200 at the inbound leg...
Is that the eventual purpose of this patch:
http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch ?
If so, do you know an equivalent for recent Asterisk version?
Thanks a lot in avance for giving the best practices around those points!
Cheers,
Florent
--
_
-- 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] RPM updates
- Original Message - From: Steven Howes steve-li...@geekinter.net On 28 Jan 2013, at 13:55, Steven Howes wrote: Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers for the replies regarding alternative repos. I'm looking to keep using the Digium ones, but they're still broken. Guess I'll just have to wait until someone at Digium notices :S I'm not involved in the build process for RPMs, but it sounds like they are waiting on the dahdi-linux 2.6.2 release to finish the new set of RPMS. I'd throw out an estimate of 1-3 weeks. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com -- _ -- 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] TLS
Hi, I tried it the implementation of TLS in asterisk 1.8.4.3 on ubuntu 10.04. I follow the tutorial: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial. and I use blink as a softphone in ny client in windows. for regular communication process (without TLS) smoothly, but when it just follow the tutorial, it is always error on his softphone: transport error. my configuration like this: certificate for the server : . /ast_tls_cert -C 10.4.71.27 -O My Super Company -d /etc/asterisk/keys certificate for Client 1 : . /ast_tls_cert -m client -c /etc/asterisk/keys/ca.crt -k /etc/asterisk/keys/ca.key -C 10.4.71.24 -O My Super Company -d /etc/ asterisk/keys -o 1001 certificate for Client 2 : . /ast_tls_cert -m client -c /etc/asterisk/keys/ca.crt -k /etc/asterisk/keys/ca.key -C 10.4.71.23 -O My Super Company -d /etc/ asterisk/keys -o 1002 sip.conf: [general] context = default udpbindaddr = 0.0.0.0 tcpenable = no tcpbindaddr = 0.0.0.0 allowguest = no allow = ulaw allow = alaw allow = gsm allow = g722 tlsenable = yes tlsbindaddr = 0.0.0.0 tlscertfile = / etc / asterisk / keys / asterisk.pem tlscafile = / etc / asterisk / keys / ca.crt tlscipher = ALL tlsclientmethod = TLSv1 [1001] context = default type = friend username = 1001 secret = 1000 dtmfmode = rfc2833 callerid = 1001 host = dynamic transport = tls [1002] context = default type = friend username = 1002 secret = 1002 dtmfmode = rfc2833 host = dynamic transport = tls extensions.conf: [general] static = yes WriteProtect = no [default] exten = 1001.1, Dial (SIP/1001, 30, tr) exten = 1001.2, Hangup exten = 1002.1, Dial (SIP/1002, 30, tr) exten = 1002.2, Hangup anyone know where's my mistake? thanks. -- _ -- 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] RPM updates
On 6 Feb 2013, at 20:06, Rusty Newton wrote: - Original Message - From: Steven Howes steve-li...@geekinter.net On 28 Jan 2013, at 13:55, Steven Howes wrote: Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers for the replies regarding alternative repos. I'm looking to keep using the Digium ones, but they're still broken. Guess I'll just have to wait until someone at Digium notices :S I'm not involved in the build process for RPMs, but it sounds like they are waiting on the dahdi-linux 2.6.2 release to finish the new set of RPMS. I'd throw out an estimate of 1-3 weeks. Hi Rusty, thanks for the update. Sounds like it's being done to save wasting time building both the old and new dahdi-linux against the new kernel. I can see why that might be done. Makes it a bit awkward if we cant build a PBX by just yum installing for a few weeks. I'll have to try rolling back to an out of date kernel on the box involved - my project is on around the 4 week mark so it'd be cutting it a little fine otherwise. Cheers for the update. Steve -- _ -- 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] Problem using ast_tls_cert script
hi daniel, are you sure the command in debian and ubuntu same? On Wed, Feb 6, 2013 at 10:59 PM, Daniel - Asterisk earohua...@gmail.comwrote: Hi List, I'm trying to set my Asterisk 1.8.20.1 with TLS on CentOS 5.9, it was easy and straightforward with Debian 6.0.6, but when I introduce this command on CentOS: #./ast_tls_cert -C 10.200.108.17 -O MyCompany -d /etc/asterisk/keys/ I got this error message: hostname: Unknown host Same result happens when using server's hostname: #./ast_tls_cert -C ast-centos -O MyCompany -d /etc/asterisk/keys/ Where 'ast-centos' is the result of 'uname -n' I've followed instructions from: http://goalbound.blogspot.com/2012/05/configure-asterisk-18110-on-centos-55.html and https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Any hint would be appreciated! Elder D. Arohuanca DCAP Lima - Peru -- _ -- 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] Somewhat OT: Specific SIP packets can cause ethernet controller reset
On 7/02/2013, at 5:08 AM, Kristian Kielhofner k...@kriskinc.com wrote: While not strictly Asterisk related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html A fascinating write-up Kristian. The information it's making it's way around the e-mail underbelly of the internet very quickly, and a lot of interested (and some very switched on) eyes are seeing it in a short amount of time, or so it seems. I first heard about it through a geek friend of mine here in NZ before I saw your posting on this list for example. Good on you for making your research results ( method) public. Well done. Pete Mundy smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] FW: How configure asterisk server extension.conf.
Best Regards,Sakharam Thorat. From: sakharam.tho...@einfochips.com To: asterisk-users@lists.digium.com Subject: How configure asterisk server extension.conf. Date: Thu, 24 Jan 2013 15:41:33 +0530 Hi, I have to create scenario like following, I have 2 sip soft phone.I configured Asterisk server on local network, on Linux.With two soft-phone , local asterisk sever, i able to communicate.Now i have communicate with other network SIP client.For that i have opened account at @sip2sip.info, they provided me credentials.Then i registered one SIP phone to local Asterisk sever and another to Sip2sip.info , Can i able to communicate with this scenario ? How i should configure extention.conf in local asterisk sever to communicate with soft-phone which registered at Sip2sip.info ??Or if you have any other idea to crate such scenario please let me know ??Please also recommend me any good SIP Developer group ?? Best Regards,Sakharam Thorat. -- _ -- 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
