Re: [asterisk-users] Asterisk 11.19.0 Now Available
On Sun, Aug 9, 2015 at 11:46 AM, Matthew Jordan mjor...@digium.com wrote: On Sat, Aug 8, 2015 at 8:26 AM, Administrator TOOTAI ad...@tootai.net wrote: Le 07/08/2015 23:54, Asterisk Development Team a écrit : The Asterisk Development Team has announced the release of Asterisk 11.19.0. [...] Hello, We have problem with patches since 11.18.0 We have to download the full tar.gz to get last version :-(. Before this, since ages, we used to patch the previous version like #patch -p0 ../asterisk-11.17.0-patch (applied to the current asterisk-11.16-0 directory), compile and install. That's all, servers where uptodate, job done. Taking a look at the header from asterisk-11.17.0-patch (and previous) we see --- asterisk-11.16.0-summary.html (.../11.16.0) (revision 433916) +++ asterisk-11.16.0-summary.html (.../11.17.0) (revision 433916) which is, diff between asterisk-11.16.0 and -in this case- the new asterisk-11.17.0 Now, since 11.18.0 version, patch is looking like diff --git a/.version b/.version index cde331b..3644f46 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -11.19.0-rc1 \ No newline at end of file +11.19.0 \ No newline at end of file As you can see patch is build against 11.19.0-rc1, not 11.18.0 How can we apply this patch to a legacy asterisk-11.18.0 tar.gz ? Thanks for any hint. That's a bug in the release scripts, which had to be rewritten when we moved to Git. We'll try to get it sorted out for the next release. In addition for the git patches you will need to use -p1 instead of -p0. 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
Re: [asterisk-users] Siren7 for Asterisk 13.5
Richard Kenner wrote: What is the proper version of the Siren7 codec to use for Asterisk 13.5.0? Since there's nothing later, does the version for 12.0 work? A Siren codec is not currently available and the one for 12 will not work. I have no timeframe for when this might change. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk queue - skills based routing (patch updated)
Le 2015-08-10 13:54, Marek Cervenka a écrit : Dne 6.8.2015 v 21:00 Sylvain Boily napsal(a): Hello, Le 2015-08-06 09:24, Marek Cervenka a écrit : hi, there is updated skills based routing patch for asterisk queue please test if you have time https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 You can find the latest version we maintain here : https://github.com/xivo-pbx/asterisk/blob/master/debian/patches/xivo_skill_queues (asterisk 13.5) We originally wrote this patch for xivo and it's included by default. Sylvain that's great! do you have plan merge it to the asterisk master? At the astricondev 2012, there was a decision to not merged this patch on app_queue because nobody really wanted to add new features. So, no there is no plan to merge this patch on the master, but we maintain it on xivo with the latest asterisk version and if someone want to work with us and people would like this patch into the master, we will be enjoy to contribute. Sylvain -- _ -- 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] Siren7 for Asterisk 13.5
A Siren codec is not currently available and the one for 12 will not work. I have no timeframe for when this might change. So the only option is to build one from the Polycom sources? I'm already doing this for Siren14 (I forget why). -- _ -- 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] Siren7 for Asterisk 13.5
Richard Kenner wrote: Alas, until we get off our butts, yes. Sorry about that. Really, we're putting as much effort into fixing things and issues that affect a lot of people. While siren7/siren14/silk are nice, there aren't as many people using them as other affected things at this moment. Is there something nontrivial that needs to be done here other than just recompiling/linking? If so, then I'm likely to run into it as well. The way formats and codecs are defined within Asterisk was changed, as a result code changes are required. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siren7 for Asterisk 13.5
On Mon, Aug 10, 2015 at 10:38 AM, Richard Kenner ken...@gnat.com wrote: A Siren codec is not currently available and the one for 12 will not work. I have no timeframe for when this might change. So the only option is to build one from the Polycom sources? I'm already doing this for Siren14 (I forget why). Alas, until we get off our butts, yes. Sorry about that. Really, we're putting as much effort into fixing things and issues that affect a lot of people. While siren7/siren14/silk are nice, there aren't as many people using them as other affected things at this moment. -- Matthew Jordan Digium, Inc. | Director of Technology 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
Re: [asterisk-users] Siren7 for Asterisk 13.5
Alas, until we get off our butts, yes. Sorry about that. Really, we're putting as much effort into fixing things and issues that affect a lot of people. While siren7/siren14/silk are nice, there aren't as many people using them as other affected things at this moment. Is there something nontrivial that needs to be done here other than just recompiling/linking? If so, then I'm likely to run into it as well. -- _ -- 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] webrtc no audio
Marek Cervenka wrote: hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) BUT call from person2(chrome) to person 3(Jitsi sip client) - works! any tips howto find the problem? You would need to look at the ICE negotiation to see if it tried and failed. After that would be looking at the DTLS negotiation. Asterisk console output could provide some information. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] load-balancing AMI and load-balancing FastAGI?
Hi, I am starting a new project to develop a predictive dialler system. - Agents can start receiving calls from the queue if agent press Available button on the browser which will unpause the queue on Asterisk. - About 100-150 concurrents calls on a Asterisk box - Call-out initiated. Other end answers. Passes AMD. Lands in Queue and direct to agents that is available and call is recorded. - Update state of the call (Ringing, Talking, etc) on the database. - Listen the events such as Hang Up from customer, check if call is successfully originated or what the failure, etc. - Agent will have ability to transfer customer call to other agent or external number. As described above to develop a predictive dialler system, is it best to use AMI or FastAGI? I am aware that I can setup FastAGI load balancing such as agitator (FastAGI reverse proxy). AMI case: load-balances incoming events/response across multiple processes (multiple AMI connections on the same asterisk machine), should the ami events/response should be pushed into RabbitMQ so the proess can read from RabbitMQ ? Thanks Paul -- _ -- 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 RealTime Sippeers, rtcachefriends=yes, phones lose registration
Hello, we have an issue where after a couple of days, a few random phones
will lose registration. I don't notice any particular pattern. Out of 200,
only about 5-10 will be suffering at any given time and we won't know until
the user complains they are not receiving calls. sip show peers does not
show the phone in the list. I see packets coming into the server, but
asterisk is not responding. Rebooting the phone to signal another REGISTER,
yields no results. It seems Asterisk simply ignores the packets. I don't
see anything in the logs that are relevant during the registration attempt.
The only thing that seems to bring the phones instantly online (no matter
how long they are in this state) is to run either 'sip reload' (not
recommended) or 'sip realtime prune peer ' and then 'sip show peer
load'. This will bring the phone instantly back up and able to receive
calls.
I never had this issue with flat files on the same server.
I noticed there was some other threads with similar traits but I don't know
if they are related. They were playing with the ignoreregexpire=yes
parameter.
I have in the sip.conf file:
rtcachefriends=yes
rtsavesysname=no
rtupdate=no
rtautoclear=no
ignoreregexpire=yes
Running asterisk 11.12.1
A sample entry in the sippeers table:
INSERT INTO sippeers
(name,defaultuser,host,secret,fromdomain,mailbox,context,nat,qualify,directmedia)
VALUES
('7293','7293','dynamic','password','ip-of-pbx','7293','from-exten','force_rport,comedia','yes','no');
Thanks!
Caesar
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Re: [asterisk-users] webrtc no audio
I'm having the same issue! The difference in my case is Asterisk server has
a public IPv4 and the browser is behind a single NAT.
I'm forwarding my configuration below (which I posted previously on
asterisk-users).
How can we debug ICE negotiation?
-- Forwarded message --
From: Vinicius Fontes vinic...@aittelecom.com.br
Date: 2015-07-27 13:54 GMT-03:00
Subject: No audio on SIP over WebRTC
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
I'm following this tutorial (
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) to
deploy WebRTC support but I'm having an issue with RTP when the WebRTC
softphone is behind NAT.
In my scenario, the Asterisk server is running a public IPv4, and the
softphone is behind NAT. I can register and make a call normally, but I
don't get any audio in neither way (Asterisk/softphone and
softphone/Asterisk). Using the very same config files but having the
softphone and Asterisk on the same network it works fine.
Any tips on how to solve this? Here's my relevant files.
*;sip.conf:*
[general]
udpbindaddr=0.0.0.0:5060
realm=10.201.0.106 ;replace with your Asterisk server public IP address or
host
transport=udp,ws,wss
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
[6000]
host=dynamic
secret=mysecret
context=default
type=friend
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
qualify=yes
[6001]
host=dynamic
secret=mysecret
context=default
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
*extensions.conf:*
[default]
exten = _6XXX,1,Dial(SIP/${EXTEN})
*rtp.conf:*
[general]
rtpstart=1
rtpend=2
icesupport=yes
stunaddr=stun.l.google.com:19302
2015-08-10 12:35 GMT-03:00 Joshua Colp jc...@digium.com:
Marek Cervenka wrote:
hello,
i'm facing strange problem
asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230
person1 to person3 are behind different NATs
audio devices double checked
call from person1(chrome) to person2(chrome) works
call from person1(chrome) to person 3(chrome) - no audio on both side
(RTP flowing only in one direction)
call from person2(chrome) to person 3(chrome) - no audio on both side
(RTP flowing only in one direction)
BUT
call from person2(chrome) to person 3(Jitsi sip client) - works!
any tips howto find the problem?
You would need to look at the ICE negotiation to see if it tried and
failed. After that would be looking at the DTLS negotiation. Asterisk
console output could provide some information.
--
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] webrtc no audio
hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) BUT call from person2(chrome) to person 3(Jitsi sip client) - works! any tips howto find the problem? -- --- Marek Cervenka === -- _ -- 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] Modifying CDR values from a hangup extension in Asterisk 13
On Tue, Aug 4, 2015 at 9:16 AM, Filip Jenicek fjeni...@kerio.com wrote:
With endbeforehexten=no I actually get two CDR entries. One for the call
and a second one for the h extension.
,13,10,sip-locals,13
13,SIP/13-0006,SIP/10-0007,Dial,SIP/10,2015-08-04
06:28:44,2015-08-04 06:28:45,2015-08-04
06:28:47,3,1,ANSWERED,DOCUMENTATION,1438669724.6,empty
,13,h,sip-locals,13
13,SIP/13-0006,,NoOp,changed,2015-08-04
06:28:47,2015-08-04 06:28:47,2015-08-04
06:28:47,0,0,ANSWERED,DOCUMENTATION,1438669724.6,changed
The first one contains the call itself. There are durations, CDR variables
set during the call, etc.
The second one contains only things configured in the h extension.
With endbeforehexten=yes, the cdr contains:
,13,10,sip-locals,13
13,SIP/13-0006,SIP/10-0007,Dial,SIP/10,2015-08-04
06:28:44,2015-08-04 06:28:45,2015-08-04
06:28:47,3,1,ANSWERED,DOCUMENTATION,1438669724.6,empty
There is only the call, nothing from the h extension.
I forgot to mention that I'm using Asterisk 13.1-cert2. Modifying CDR
records in the h extension used to work fine with Asterisk 1.8.
By analyzing the code I must confirm that the endbeforehexten option
behaves exactly according to its description:
As each CDR for a channel is finished, its end time is updated
and the CDR is finalized. When a channel is hung up and hangup
logic is present (in the form of a hangup handler or the
literalh/literal extension), a new CDR is generated for the
channel. Any statistics are gathered from this new CDR. By enabling
this option, no new CDR is created for the dialplan logic that is
executed in literalh/literal extensions or attached hangup handler
subroutines. The default value is literalyes/literal, indicating
that a CDR will be generated during hangup logic./para
I tried to delay the h extension by several seconds and I found out,
that the CDR record is sent to the cdr backend later. Unfortunately, it is
not modifiable from the h extension, because the cdr_object is already in
the finalized table.
Is there a way how to modify the CDR without hacking the code?
Unfortunately, no.
How bad idea is it to comment the (it_cdr-fn_table ==
finalized_state_fn_table) tests in ast_cdr_setuserfield and ast_cdr_setvar
and thus allow the h extension write to a finalized CDR?
Well... I'm not sure :-)
As the guy who signed himself up for the dubious honour of porting the CDR
code to Asterisk 13 - and trying to figure out a consistent way to make it
work - I err'd on the side of extreme caution. That is, if someone could
make a mess of things, I should probably try to keep it from happening.
A CDR can be finalized in a variety of ways:
- Due to someone leaving a bridge
- Due to a channel being hung up
- Due to the CDR being forked
Of those, modifying values is generally dangerous only in the fork
scenario, as it may result in a CDR that a user 'ended' being modified.
This is a concern when, as updating a value on a CDR walks the entire chain
of CDRs, for all CDRs related to the channel:
for (; (cdr = ao2_iterator_next(it_cdrs)); ao2_unlock(cdr),
ao2_cleanup(cdr)) {
ao2_lock(cdr);
for (it_cdr = cdr; it_cdr; it_cdr = it_cdr-next) {
struct varshead *headp = NULL;
if (it_cdr-fn_table == finalized_state_fn_table) {
continue;
}
if (!strcasecmp(channel_name, it_cdr-party_a.snapshot-name)) {
headp = it_cdr-party_a.variables;
} else if (it_cdr-party_b.snapshot
!strcasecmp(channel_name,
it_cdr-party_b.snapshot-name)) {
headp = it_cdr-party_b.variables;
}
if (headp) {
set_variable(headp, name, value);
}
}
}
ao2_iterator_destroy(it_cdrs);
Currently, the fact that the CDR is in the finalized state is what prevents
that value from being updated on CDRs that are effectively closed.
Now, all of that being said: this is one of those cases where the current
behaviour - which is handling an extreme edge case - feels worse than
ignoring that edge case. It's not like we let folks update core CDR
values in any case, so you aren't in any danger of changing the billsec on
a forked CDR. The worst that happens is you update the userfield on forked
closed CDRs when you didn't think it would update, in which case I
suppose you could just use another field. Or read it first and append it
from the dialplan.
Is there any chance the feature was left out by an accident and if so, is
there a plan to add it again?
My extensions.conf:
exten = h,1,NoOp(${CDR(userfield)})
exten = h,n,Set(CDR(userfield)=changed)
exten = h,n,NoOp(${CDR(userfield)})
exten = h,n,System(sleep 5)
exten = h,n,NoOp(${CDR(userfield)})
exten = 10,1,Set(CDR(userfield)=empty)
exten = 10,n,Dial(SIP/10)
Detailed log:
http://pastebin.com/fZ9RAhL4
I'd be fine if you'd like to open an issue for it. If you have a
Re: [asterisk-users] asterisk queue - skills based routing (patch updated)
Dne 6.8.2015 v 21:00 Sylvain Boily napsal(a): Hello, Le 2015-08-06 09:24, Marek Cervenka a écrit : hi, there is updated skills based routing patch for asterisk queue please test if you have time https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 You can find the latest version we maintain here : https://github.com/xivo-pbx/asterisk/blob/master/debian/patches/xivo_skill_queues (asterisk 13.5) We originally wrote this patch for xivo and it's included by default. Sylvain that's great! do you have plan merge it to the asterisk master? -- --- Marek Cervenka === -- _ -- 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
