Re: [asterisk-users] switching from simple_bridge technology to native_rtp issue
On Thu, Jul 10, 2014 at 4:28 AM, Sameer Rathod wrote: > Hi Matt, > > I also tested the directmedia=yes over 3g connection ie with a public ip but > I am getting only one way audio > am I doing anything wrong? > If you are getting one way audio when direct media is enabled, then one of the devices cannot find the other. This is most likely because one of the devices is behind a NAT. If both devices are truly publicly accessible, then you would have to look at a pcap at the re-INVITEs sent to the devices to determine where Asterisk told the devices to send their media, then debug your network to determine why media could not be sent directly between those two devices. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] switching from simple_bridge technology to native_rtp issue
On Wed, Jul 9, 2014 at 4:56 AM, Sameer Rathod wrote:
> Hi,
>
> with canreinvite=no and directmedia=no I and getting the message in the logs
> for all calls
>
> "switching from simple_bridge technology to native_rtp"
>
>
> -- Executing [102@mkg:1] Dial("SIP/101-0017", "SIP/102") in new stack
> == Using SIP RTP CoS mark 5
> -- Called SIP/102
> -- SIP/102-0018 is ringing
> -- SIP/102-0018 answered SIP/101-0017
> -- Channel SIP/101-0017 joined 'simple_bridge' basic-bridge
> <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
> -- Channel SIP/102-0018 joined 'simple_bridge' basic-bridge
> <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
>> Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from
> simple_bridge technology to native_rtp
>> 0x7f427c068a10 -- Probation passed - setting RTP source address to
> 111.118.250.236:49344
>> 0x7f427c068a10 -- Probation passed - setting RTP source address to
> 111.118.250.236:49344
>> 0x7f42500168d0 -- Probation passed - setting RTP source address to
> 111.118.250.236:26326
>> 0x7f42500168d0 -- Probation passed - setting RTP source address to
> 111.118.250.236:26326
> -- Channel SIP/101-0017 left 'native_rtp' basic-bridge
> <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
> -- Channel SIP/102-0018 left 'native_rtp' basic-bridge
> <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
> == Spawn extension (mkg, 102, 1) exited non-zero on 'SIP/101-0017'
>
>
>
> I cannot understand why asterisk state diff bridges if all works same
>
> please can anyone explain me the working bridging concept and how to
> configure and use bridges to route the rtp externally form asterisk.
>
I think I just answered this in your other thread, but I'll repeat it here.
First, canreinvite has been deprecated as a naming convention for ...
a long time. It's not even documented any more. The code will accept
it, but all you're doing is setting the directmedia option twice:
} else if (!strcasecmp(v->name, "directmedia") ||
!strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
The native RTP bridge in Asterisk 12 manages bridges between two RTP
capable channels. The bridge can either be formed remotely (in which
case the media flows between the endpoints) or locally, in which case
the media is swapped across the ports. It will attempt to perform a
remote bridge if possible, while falling back to a local bridge if a
remote bridge is not possible.
In your particular case, you've explicitly told it to *not* do
directmedia. So it won't perform a remote bridge.
Even if you set directmedia=yes (or one of its variants), you may not
have a successful remote bridge if one of the endpoints is behind a
NAT. The sip.conf sample configuration documentation is actually quite
good on this subject:
;--- MEDIA HANDLING
; By default, Asterisk tries to re-invite media streams to an optimal
path. If there's
; no reason for Asterisk to stay in the media path, the media will be
redirected.
; This does not really work well in the case where Asterisk is outside and the
; clients are on the inside of a NAT. In that case, you want to set
directmedia=nonat.
;
;directmedia=yes; Asterisk by default tries to redirect the
; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of
them is behind a NAT).
; The default setting is YES. If you
have all clients
; behind a NAT, or for some other
reason want Asterisk to
; stay in the audio path, you may want
to turn this off.
; This setting also affect direct RTP
; at call setup (a new feature in 1.4
- setting up the
; call directly between the endpoints
instead of sending
; a re-INVITE).
; Additionally this option does not
disable all reINVITE operations.
; It only controls Asterisk generating
reINVITEs for the specific
; purpose of setting up a direct media
path. If a reINVITE is
; needed to switch a media stream to
inactive (when placed on
; hold) or to
Re: [asterisk-users] packet2packet bridging
On Wed, Jul 9, 2014 at 2:47 AM, Sameer Rathod wrote: > Hi, > > Please clear me on this topic I am confused > > My log show "switching to native rtp". > Did this line means that the audio is not coming to the asterisk server any > more and asterisk only send the re- invite packet to both the clients ? > > Am I right or wrong ? > You are wrong (sorry). All that means is that the bridging has switched to a native RTP bridge. That bridge comes in two variants: a local packet to packet bridge (where the media flows through Asterisk but is not decoded - RTP is merely swapped between ports) and a remote bridge. The remote bridge is where the two channels are in a bridge in Asterisk, but media flows directly between the endpoints. If your endpoints are behind a NAT, then no, you cannot use a remote bridge. No amount of hoping or tinkering will make it so. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk crashes when reloading configs...
On Wed, Jul 2, 2014 at 12:38 PM, Carlos Chavez wrote: > I am having a very strange problem. We use Asterisk 11.X (have tried > several versions, including certified) which reads its config files in > realtime from a SQLITE3 database. Everything runs fine but lately asterisk > has been crashing when we issue a "reload" command via Manager. Our web > interface uses AMI to reload the dialplan and right after it does that ( I > can see the results on the CLI) asterisk crashes. This does not seem to > happen every time but some days it crashes often. Any ideas where to start > looking for the problem? > Please get a backtrace illustrating the problem: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace Once you have a properly generated backtrace, open an issue on issues.asterisk.org. Thanks - Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] PJSIP question
On Wed, Jun 18, 2014 at 6:05 AM, CDR wrote: > A few months ago I started using and had to abandon PJSIP because my > dialplan could not read the inbound signalling IP address, which I can > read now in Asterisk11 using CHANNEL(recvip). My app relies on this > information. The > question is, is it possible now access the signalling IP of an > incoming SIP call using PJSIP? > Philip > The CHANNEL function [1] was integrated with chan_pjsip in the first official release, 12.0.0. You can obtain the address of the remote party using CHANNEL(pjsip,remote_addr). [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_CHANNEL -- Matthew Jordan Digium, Inc. | Engineering Manager 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] WSS over Asterisk
On Wed, Jun 11, 2014 at 1:32 PM, William Hetherington wrote: > Chrome 35 broke all of this you need to be using DTLS now I believe. > > I had working secure web sockets with asterisk 12.2.x and chrome 34 > and then google broke eveything :) > > I have not yet got around to test out DTLS etc. with chrome 35 > > Just so I don't waste too much time when I go to test, does anyone know if > all that's required for DTLS on the asterisk side is the following in > sip.conf? > > dtlsenable=yes > dtlsverify=yes > dtlsrekey=60 > dtlscafile=/usr/local/share/ca-certificates/myCA.crt > dtlscertfile=/etc/ssl/mycert.com.pem > dtlssetup=actpass > > I assume I also need TLS configs in http.conf > > Signalling is independent of the media; DTLS only affects the media. However, there are known issues with Chrome's negotiation of DTLS and Asterisk - see https://issues.asterisk.org/jira/browse/ASTERISK-22961 -- Matthew Jordan Digium, Inc. | Engineering Manager 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] WSS over Asterisk
On Wed, Jun 11, 2014 at 2:58 AM, Steve Ng wrote: > Hi, > > Have anyone tried using SIPML5 to connect to Asterisk over wss? > > I'm having the error as shown below > > Connecting to 'wss://54.xxx.xxx.xxx:8080/ws' SIPml-api.js?svn=224:1 > ==stack event = starting SIPml-api.js?svn=224:1 > __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 > __tsip_transport_ws_onclose SIPml-api.js?svn=224:1 > ==stack event = failed_to_start > > > Where if I'm connecting through ws://54.xxx.xxx.:8080/ws, it works > fine. Any idea why? > > There was a bug in secure WebSockets (tracked under ASTERISK-21930) that was fixed in Asterisk 11.9.0: http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-11.9.0-summary.html Which version of Asterisk are you using? Is it 11.9.0 or later? -- Matthew Jordan Digium, Inc. | Engineering Manager 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] CDR custom variable on second call leg - via originate or .call file
On Wed, Jun 11, 2014 at 9:10 AM, Tiago Geada wrote:
> Hi,
>
> Let me append some extra info
>
> cdr variable foo, shows on database, but value 'bar' doens't
>
> its not even shown in the insert query
>
> I tried with master_channel but no change
>
>
I think you need to be a bit more specific about what CDR records you're
getting and what you'd like to have happen.
You have the following call file:
>
>>
>>
>>
>> ## test call file
>>
>>
>>
>> Channel: Local/queue@TiagoGeada
>>
>> CallerID: teste-geada:0:210332450:
>>
>> MaxRetries: 0
>>
>> RetryTime: 1
>>
>> WaitTime: 8640
>>
>> Account: teste-geada
>>
>> Context: TiagoGeada
>>
>> Extension: outbound
>>
>> Archive: Yes
>>
>>
>>
>>
>>
>
This will create a Local channel with two halves. The ;2 half will execute
in the dialplan at TiagoGeada,queue,1 - the ;1 half will execute in the
dialplan at TiagoGeada,outbound,1. The ;2 Local channel will execute first
until it is Answered; once Answered, that will trigger the ;1 half to start
execution. That will create two CDRs, one for each Local channel half.
MASTER_CHANNEL won't apply here, as MASTER_CHANNEL only applies to a
Parent/Child relationship between channels, that is, when one channel has
created another channel. This occurs when a channel dials another channel.
The ;1 side didn't create the ;2 side, they are effectively two sides of
the same "channel".
>
>>
>>
>>
>> ## dialplan
>>
>>
>>
>> queue => {
>>
>> Set(CDR(remoteUid)=foo);
>>
>> Queue(TiagoGeada,t,,,100);
>>
>> Hangup();
>>
>> }
>>
>>
>>
>> outbound => {
>>
>> //NoCDR();
>>
>> //ForkCDR(vdD);
>>
>> //ResetCDR(v);
>>
>> Set(CDR(remoteUid,r)=bar);
>>
>> Dial(Local/932485457@outbound,,gT);
>>
>> Hangup();
>>
>> }
>>
>>
>>
Looking at your Dialplan for the outbound extension, you dial yet another
Local channel. I would expect this to result in 3 CDR entries:
Source Channel Destination Channel
Local/queue@TiagoGeada;2
Local/queue@TiagoGeada;1 Local/932485427@outbound;1
Local/932485457@outbound;2
So, the question is, which CDR are you talking about? What value do you
want where? Keep in mind that unless all channels are answered, they won't
show up in your CDRs (unless you have unanswered=yes set in cdr.conf).
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
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Re: [asterisk-users] Hold
On Wed, Jun 11, 2014 at 12:45 PM, jonathan white wrote:
> Can you write the unique variable to astdb and then write it back to the
> variable?
>
> Not sure I have thought this through
>
> J
> On 11 Jun 2014 18:42, "Kelly Opal" wrote:
>
>> Hi
>> I am trying to set up a hold system so that a call is always
>> parked in the same spot no matter how many times it is picked up. My
>> problem is I cannot fins a variable the identifies the call all the way
>> through until it is destroyed. ${UNIQUEID} and ${CHANNEL} both seam to get
>> lost when the call is parked. I tried setting
>> set($[“${UNIQUEID}-hold”=”701”])
>> and
>> set($[“${CHANNEL}-hold”=”701”])
>>
>> and both work fine until I do a transfer to park. Then both variables are
>> blank. Is there any variable that is persistent to a call through all of
>> the transfers.
>>
>> asterisk 11.6-cert1
>> centos 5.7
>>
>> Thanks
>>
>> Kelly
>>
>>
You are most likely running into masquerades. A masquerade is an internal
operation in Asterisk that involves renaming a channel. When this occurs,
your AMI client will receive a sequence of Masquerade and Rename events.
Your client will need to update its tracking of the channel based on those
events.
Alternatively, you can move to Asterisk 12. One of the major projects that
was done in that version was to remove the visibility of masquerades from
external systems (and mostly purge them internally), such that channels
have a stable, consistent identifier for the channel throughout its
lifetime.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
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 --
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Re: [asterisk-users] SSL/TLS weakness impact on Asterisk authentication
On Tue, Jun 10, 2014 at 4:44 PM, Michelle Dupuis wrote: > After reading about the 2 major SSL (and TLS?) weaknesses discovered > this year, I was wondering how it affects asterisk. > Asterisk uses OpenSSL for TLS. So, the answer is, it depends on the version of OpenSSL that was installed for your Asterisk server. See http://blogs.digium.com/2014/04/11/asterisk-heartbleed/ for more information. > Does the SIP authentication use TLS - or something that was recently > broken? Is there a risk of exposing passwords? > SIP signalling - in both chan_sip and chan_pjsip - can use TLS as a transport. If your OpenSSL version is one of those affected by the various vulnerabilities, then yes, you are at risk. This also applies to all other modules in Asterisk that use TLS, including AMI, the HTTP server, and others. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] 'restart when convenient'
On Wed, May 28, 2014 at 1:08 PM, Matthew Jordan wrote: > On Wed, May 28, 2014 at 12:47 PM, Doug Lytle wrote: >>>> Perhaps i should join the -dev list to find out what 'convenient' >>>> actually means for the process... >> >> The dev list is for discussions of coding Asterisk, this would be considered >> a user's question. Many of the devs are subscribed to this list as well. >> > > The various 'stop/restart' flavours and what they mean: > > * convenient - wait until all channels have hung up. If no channels > are made, keep waiting. Once all channels have hung up, and no new > channels are made, sneak in and ask all of the modules shut down and > to clean up after themselves - including waiting for all CDRs to get > written. Apologies, that should read "If new channels are made, keep waiting" -- Matthew Jordan Digium, Inc. | Engineering Manager 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] 'restart when convenient'
On Wed, May 28, 2014 at 12:47 PM, Doug Lytle wrote: >>> Perhaps i should join the -dev list to find out what 'convenient' >>> actually means for the process... > > The dev list is for discussions of coding Asterisk, this would be considered > a user's question. Many of the devs are subscribed to this list as well. > The various 'stop/restart' flavours and what they mean: * now - tell all CDRs to go submit themselves. Tell all channels to hang up, and wait up to 15 seconds for all channels to go away. Don't allow new channels to be made. After 15 seconds, shut down anyway. Clean up of resources in modules is "best effort" - don't wait for anything to get written or otherwise dispose of itself. * gracefully - tell all CDRs to go submit themselves. Wait until all channels have hung up, but don't hang them up. Don't allow new channels to be made. Once all channels have hung up, ask all of the modules to clean up after themselves - including waiting for CDRs to get written. * convenient - wait until all channels have hung up. If no channels are made, keep waiting. Once all channels have hung up, and no new channels are made, sneak in and ask all of the modules shut down and to clean up after themselves - including waiting for all CDRs to get written. Note that while one shutdown command is executing, another cannot be started. Thus, if you issue a 'core stop gracefully' and decide later to escalate it to 'core stop now', by default, Asterisk will refuse the 'core stop now' command. You can, however, stop the 'core stop gracefully' by issuing 'core abort shutdown', which will cause Asterisk to stop the existing shutdown attempt and return to normal processing. You can then issue another shutdown command at your leisure. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] BLF and notifyringing in Asterisk 11
On Fri, May 23, 2014 at 4:51 PM, Kevin Larsen wrote: > I am trying to get something working that is just not doing quite what I > want. It may not be possible, but I figured it was worth asking about. > > The details: > Asterisk 11.6.0 > Polycom SoundPoint IP650 phones running 4.03 firmware. > > We have a queue with 4 phones in it. ringinuse is set to yes and the stategy > is ringall. In sip.conf, we have notifyringing set to yes as well. Asterisk > is sending messages of the type application/dialog-info+xml to the phones. > > This works nicely in almost every scenario. We have one person on the queue > who answers the phones first, the rest of us only pick up if he is already > on another call and not picking up. We have ringinuse set to yes because > there are many times that he will be able to end his current call to pick up > the new one, so we want to keep this setting. However, if a call comes into > the queue and he is on another line, we would like his blf light to stay at > the inuse value (red on the polycom) and not the ringing value (flashing > green on the polycoms). > > Now the problem. If I set notifyringing=no on the sip definition for his > extension, it doesn't seem to get applied. If I set notifyringing=no in the > general section, then it does get applied. However, if I put it in the > general section, then none of the phones in my queue ever show a ringing > state. When they are ringing, they show the solid red light of the in use > state. > > What I would like is to see the following happen: > If no one is on a call, all phones show ringing on their respective BLFs. > If one phone is on a call and a second comes in, the phone on the call stays > in use (solid red) and the rest show ringing (flashing green). So far, no > matter what combinations of notifyringing I use, I can only get either all > the phones to show ringing or all of them to show in use. > > The state being sent to the polycom is 'early' for a ringing phone and > 'confirmed' for an in use phone. In the case of a phone that is both in use > and ringing, I get a state of 'confirmed' followed immediately by a state of > 'early'. This is all with notifyringing set to yes. If I read the > description of notify ringing correctly in the sample sip.conf file, it > seems like setting it to no should work, but it does not. > > ;notifyringing = no ; Control whether subscriptions already > INUSE get sent > ; RINGING when another call is sent > (default: yes) > > Not sure if this rises to the level of a bug or is just my misunderstanding > of how this should work. With the description above, I would expect that > setting notifyringing to no would mean that I get the early state if the > phone isn't already on a call, but would remain at a confirmed state if a > second call came in while already on a call. Unfortunately, notifyringing is only set in the [general] section in sip.conf. It does not have a peer level override. It would be nice if it was set on a peer by peer basis - that would be a useful improvement. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] "transmit_silence" not properly recognized on 1.8 ?
On Sat, May 24, 2014 at 3:03 PM, Maximilian Grobecker
wrote:
> Hello,
>
> I've got a problem at the moment, that setting "transmit_silence = yes"
> seems to have no effect on Asterisk 1.8-Certified.
>
> Although it's enabled and "core show settings" confirms, that it is
> really enabled, there are no RTP packets sent by Asterisk when waiting
> for DMTF input or when "Wait()" is called.
> Also, there seems to be a small gap of 2 or 3 not sent packets when
> playing several files one after another.
>
>
Without looking at a DEBUG log, I'm not sure what anyone can tell you.
Wait does call ast_safe_sleep_conditional, which enables a silence
generator on the channel if (a) there is no generator on the channel
and (b) ast_opt_transmit_silence is true (which it should be based on
your configuration):
/* If no other generator is present, start silencegen while waiting */
if (ast_opt_transmit_silence && !chan->generatordata) {
silgen = ast_channel_start_silence_generator(chan);
}
Starting a silence generator should emit a debug level 1 message
indicating that it has started. If your log doesn't show that, then
there may be another generator present that is preventing silence from
kicking off.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
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Re: [asterisk-users] SQLite3 astdb back-end
On Fri, May 2, 2014 at 1:11 AM, thufir wrote: > How do you load the contact list? It's a database? Sqlite3? > > https://wiki.asterisk.org/wiki/display/AST/SQLite3+astdb+back-end > > I'm not clear on what this specific database does. If it's not this > specific database which has contact information, which database does? > Prior to Asterisk 10, Asterisk's internal database was a Berkeley database. For a variety of reasons, in Asterisk 10, we moved to using SQLite. Generally, this change was "under the hood", but people upgrading from Asterisk 1.8 to a later version needed to be aware of it. The database really has two purposes: (1) Asterisk uses it to store information that needs to persist between reloads/restarts. Various modules use it for this purpose automatically. (2) Users can use it to store arbitrary data. This is done using a variety of dialplan applications and functions: * https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB * https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB_DELETE * https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB_EXISTS * https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB_KEYS * https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_DBdeltree Whether or not you store 'contact information' (and that could have a variety of meanings, so I won't interpret it specifically) is up to you. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Putting a notice in the logs from the dialplan
On Thu, May 1, 2014 at 9:39 AM, Ishfaq Malik wrote: > > That works a treat, thank you. > > > On 1 May 2014 15:28, Steven Wheeler wrote: >> >> On Thu, May 1, 2014 at 8:37 AM, Ishfaq Malik wrote: >> >> Hi >> >> >> >> Using asterisk 1.8 >> >> >> >> NoOp and Verbose both put messages into the logs as VERBOSE, is there any >> way to put a message into the logs as NOTICE from within a dial plan? >> >> >> >> Thanks in advance >> >> >> >> What about the Log application? It is available on our Asterisk 1.8.26 box. >> >> >> Connected to Asterisk 1.8.26.0 >> >> Verbosity is at least 3 >> >> CLI> core show application Log >> >> >> >> -= Info about application 'Log' =- >> >> >> >> [Synopsis] >> >> Send arbitrary text to a selected log level. >> >> >> >> [Description] >> >> Sends an arbitrary text message to a selected log level. >> >> >> >> [Syntax] >> >> Log(level,message) >> >> >> >> [Arguments] >> >> level >> >> Level must be one of 'ERROR', 'WARNING', 'NOTICE', 'DEBUG', 'VERBOSE' >> >> or 'DTMF'. >> >> message >> >> Output text message. >> >> >> >> [See Also] >> >> Not available >> >> >> Ha! Just when you think you've found every corner of Asterisk, you turn around and there's something else. Just goes to show, you learn something new every day. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Wed, Apr 30, 2014 at 8:37 AM, Administrator TOOTAI wrote: > Le 30/04/2014 15:19, Matthew Jordan a écrit : > > >> On Wed, Apr 30, 2014 at 8:13 AM, Administrator TOOTAI >> > ad...@tootai.net>> wrote: >> >> Please, people from Digium, Matt again closed the new bug >> ASTERISK-23689 I opened (clone from 23683) telling that it's not a >> bug. Did he carefully read the comments on the new bug? If not, >> please forward him this email, *it's* a bug or you have to explain >> me why it is not! >> >> >> I asked you not to clone and issues and to take your issue to the mailing >> list (which you did, thank-you). Cloning issues makes a mess of the issue >> tracker, and causes information to get lost. >> >> If your issue is deemed to be a bug, the original issue will get >> re-opened. >> > > I cloned the issue as it is a bug and I could explain how to reproduce it. > If I shouldn't clone the bug, please explain me how to do to inform > developpers about new informations concerning a closed bug. > > That say, sorry for inconvenience. > > 1. Bug marshals watch the asterisk-bugs mailing list. All updates to all issues in JIRA get sent to that mailing list - even comments on closed issues. 2. Bug marshals also hang out in the #asterisk-bugs IRC channel. You can talk to a bug marshal in that channel as well. 3. Finally, we all watch the mailing lists (pretty much all of the mailing lists, no less). This is all documented on the Asterisk wiki's [1] Asterisk Issue Guidelines. There's even a checkbox when you file an issue that asks if you read the guidelines... you did read them, right? :-) [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Submittingthebugreport -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Putting a notice in the logs from the dialplan
On Thu, May 1, 2014 at 8:37 AM, Ishfaq Malik wrote: > Hi > > Using asterisk 1.8 > > NoOp and Verbose both put messages into the logs as VERBOSE, is there any > way to put a message into the logs as NOTICE from within a dial plan? > > Thanks in advance > > > Nope, unfortunately not. It would be a relatively trivial addition to add a dialplan application that could emit an Asterisk logging message at any one of the various levels, if someone were interested. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Channel names
On Thu, May 1, 2014 at 7:17 AM, Ishfaq Malik wrote: > Hi > > I'm using asterisk 1.8. > > How are channel names constructed. I always thought they were > > /- > > but I've had a lot of instances where a channel name doesn't have the > correct peer as part of it. > > Is it unwise to use channel names to extract the peers involved in a call? > > > How a channel is named is a function of the channel technology. Which channel technology(ies) are you curious about? Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] "CBAnn" channel not going away in Asterisk 12
On Thu, May 1, 2014 at 7:18 AM, Richard Kenner wrote: > > It may show up in 'bridge show all' - but I'd actually expect it not > > to show up there either. > > Actually, it does. I have a screen full of bridges with 0 channels. > > I just tried an experiment where all I have is > > exten => 329,1,Answer(1000) > same => n,Confbridge(1234) > > with absolutely nothing else going on and those leak too. I need to > understand > why I'm seeing this and nobody else is. > > Please go ahead and open an issue and attach the refs log and the full DEBUG log. That will allow us to understand what's occurring here. Thanks - Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] "CBAnn" channel not going away in Asterisk 12
On Wed, Apr 30, 2014 at 8:20 AM, Richard Kenner wrote: > > If the reference count on the bridge is off, you should see the > conference > > bridge 'hanging around' after the last participant has left. > > And how would I be sure this is the case? I did "core set debug 1" and > didn't see the debug line about destroying the conference, but it doesn't > show up in "confbridge list". > > That's not terribly surprising, as the bridge is typically removed from the list of active conferences prior to the destructor being called (the destructor being the thing that releases memory, not the thing that notifies everyone that this thing *should* be destroyed). It may show up in 'bridge show all' - but I'd actually expect it not to show up there either. Really, I think we're pretty positive there's a ref leak (since otherwise, the CBAnn channel would be long gone). If you can get a ref debug log and the standard Asterisk DEBUG log showing the problem, that would help a lot in finding out what is going on. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Wed, Apr 30, 2014 at 8:13 AM, Administrator TOOTAI wrote: > Please, people from Digium, Matt again closed the new bug ASTERISK-23689 I > opened (clone from 23683) telling that it's not a bug. Did he carefully > read the comments on the new bug? If not, please forward him this email, > *it's* a bug or you have to explain me why it is not! > > I asked you not to clone and issues and to take your issue to the mailing list (which you did, thank-you). Cloning issues makes a mess of the issue tracker, and causes information to get lost. If your issue is deemed to be a bug, the original issue will get re-opened. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] "CBAnn" channel not going away in Asterisk 12
On Tue, Apr 29, 2014 at 6:03 PM, Richard Kenner wrote:
> > If the channel still hangs around after the conference is destroyed
> > then there is a problem.
>
> Am I missing something obvious: I'm looking in the confbridge_exec
> function. I see a "conference = NULL" line, but no attempt to free
> that structure, which is what I understand will destroy the playback
> channel. So where it is freed?
>
Conferences are reference counted objects. When the reference count reaches
0 on the conference object, its destructor is automatically called. The
destructor, in this case, is destroy_conference_bridge. That is where the
CBAnn channel should be hung up.
/* Try to allocate memory for a new conference bridge, if we
fail... this won't end well. */
if (!(conference = ao2_alloc(sizeof(*conference),
destroy_conference_bridge))) {
ao2_unlock(conference_bridges);
ast_log(LOG_ERROR, "Conference '%s' could not be created.\n",
conference_name);
return NULL;
}
If the reference count on the bridge is off, you should see the conference
bridge 'hanging around' after the last participant has left. If so, please
file a bug report. We'll need a REF_DEBUG log to figure out who the guilty
party is in holding onto a reference. The easiest way to get that is to
reproduce the problem using the latest from the 12 branch (as we made
reference count debugging easier just recently). Enable REF_DEBUG in
menuselect under Compiler Flags, make/make install, and re-run the scenario
that reproduces the result. A refs file will be created in your Asterisk
log directory - attach that to the issue along with DEBUG log.
Thanks!
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
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Re: [asterisk-users] IAX2 trunk on IPV6
On Tue, Apr 29, 2014 at 1:06 AM, Xengis Khan wrote: > Hi, > I have installed asterisk-1.8.25.0 on an Ubuntu server which has both an > ipv6 ip and ipv4 ip(real ip) assigned. And I have a client ubuntu with only > ipv4 ip(local ip) installed asterisk-1.8.25.0 . I want to configure the > client asterisk with the server asterisk as IAX2 peer and want to connect > to the IPV6 ip. I bind the server with ipv6 and also sending the > registration request from the client(peer) to the ipv6 address. But its not > peering. following is the client's iax.conf > > register => peer1:peer1pass@[IPV6]:port > > [peer1] > type=peer > context=topeer > username=peer1 > secret=peer1pass > trunk=yes > host=XXX.XXX.XXX.XXX > port= > disallow=all > allow=g729:40,g723:30 > qualify=yes > > Also my confusion is what value will be in 'host' property. I assigned as > host=[IPV6]...but it shows error. > Can anyone help with this issue. > > IAX2 does not support IPv6 in that version of Asterisk. IPv6 support was added to chan_iax2 in Asterisk 12 [1]. [1] https://wiki.asterisk.org/wiki/display/AST/New+in+12 -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Does CalDAV require neon-0.29 , not 0.30?
On Sun, Apr 27, 2014 at 12:46 PM, Sean Darcy wrote:
> Asterisk-11.9.0, Fedora 20:
>
> res_calendar_caldav.so => (Asterisk CalDAV Calendar Integration)
> [Apr 27 10:49:13] ERROR[4255]: res_calendar_ews.c:911 load_module: Exchange
> Web Service calendar module require neon >= 0.29.1, but neon 0.30.0: Library
> build, IPv6, Expat 2.1.0, zlib 1.2.8, GNU TLS 3.1.13. is installed.
>
> Is this a bug, or do I need to downgrade to 0.29?
>
It is not a bug. res_calendar_ews was written to use neon 0.29. It
checks the version using ne_version_match, the documentation of which
specifies that for every version of neon beginning with major version
0, all versions of neon are assumed to be incompatible:
{quote}
The ne_version_match function returns non-zero if the library version
is not of major version major, or the minor version is less than
minor. For neon versions 0.x, every minor version is assumed to be
incompatible with every other minor version.
{quote}
http://www.webdav.org/neon/doc/html/refvers.html
You should either downgrade to 0.29, or else have a community
developer determine if res_calendar_ews is compatible with later
versions of neon.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Problem building Asterisk-12.2.0
On Sat, Apr 26, 2014 at 6:12 PM, Richard Kenner wrote: >> I think you need the libuuid and libuuid-devel packages. > > "yum list available" was not showing any such package. > > I installed a few other packages, including "uuid-dce-devel" and one of them > did the trick, but the install-prereq script wasn't good enough. > What distro are you building on? I'm running both Ubuntu 12.04 and CentOS 6.5 locally. Both have the libraries listed in install_prereq. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] how to configure callcentric peer
> > > Thanks for the response, but no: > > # grep -R host * | grep callcentric > exts/callcentric.sip.conf:host=callcentric.com > # grep -R host * | grep 204 > # > > Why in the world is asterisk trying to match on the FROM header?? > Because that's what Asterisk has always done to match an inbound request to a peer. Quoting from sip.conf.sample: ; When naming devices, make sure you understand how Asterisk matches calls ; that come in. ; 1. Asterisk checks the SIP From: address username and matches against ; names of devices with type=user ; The name is the text between square brackets [name] ; 2. Asterisk checks the From: address and matches the list of devices ; with a type=peer ; 3. Asterisk checks the IP address (and port number) that the INVITE ; was sent from and matches against any devices with type=peer -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Alembic - Asterisk 11
On Mon, Apr 14, 2014 at 12:26 PM, Josh Metzger wrote: > I've had years of experience using ODBC for CDR, SIP, and extensions with > Asterisk. One thing that has been problematic in the past is with > documentation as far as database tables changing between versions (even > within minor releases, though that was back in the 1.4 days). I was excited > to see there is a plan for better managing that on Asterisk 12 via Alembic. > All that being said, are there any plans to implement that with Asterisk 11, > since that is the current LTS release? Or are we pretty sure the table > structure won't be changing within that version through the rest of its > lifespan, making such an effort a waste? > There are no plans to implement Alembic in Asterisk 11. This is for a few reasons: (1) Asterisk 11 is an LTS and is already released. Simply introducing Alembic at this time for it is potentially disruptive. Asterisk 12, as a Standard release, was the appropriate place to introduce such a new feature. (2) Schema changes in Asterisk 11 are rare, and would only be done as a result of a critical bug that absolutely required such a change. Schema changes mid-release would be noted in the UPGRADE notes. Asterisk 12, on the other hand, has limited support for new features. Some of these new features - such as new parameters for the PJSIP stack - require new database fields. As such, various schemas have changed more frequently in this release; in this situation, Alembic is far more useful. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] AMI and pyst
On Thu, Apr 10, 2014 at 9:14 AM, Ishfaq Malik wrote: > > Does anyone on this list use pyst for AMI purposes? > > If so, can you point me in the direction of some simple examples. There seems > to be none anywhere online. Probably doesn't help that I'm not that > experienced at python but not insurmountably so. > > Thanks in Advance > > Ish > > Hey Ish - This isn't directly answering your question, but I noticed no one chimed in. At Digium we don't use pyst for Python integration with Asterisk, so I don't have any experience with it. We do, however, use starpy (https://github.com/asterisk/starpy) extensively in the Asterisk Test Suite. It does lock you into using twisted (https://twistedmatrix.com/trac/) - which has both pros and cons - but it may be a viable alternative for you if pyst doesn't work out. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk CLI Banner
On Fri, Mar 28, 2014 at 9:10 AM, Thorsten Göllner wrote: > Am 28.03.2014 10:32, schrieb Haider Khalil: > > Hello Experts, > > I want to know if there is any way to modify welcome banner on asterisk > console when I connect using "asterisk -r" > > > Hi, > > did you compile asterisk from source? Take a look at main/asterisk.c (line > 174 in asterisk v 11.5.1). I think you have to change it there manually and > recompile it. > > Please note that modifying the banner in main/asterisk.c may cause you to violate the licensing of Asterisk, specifically Section 1 of the GPL (if you distribute the modified source in any fashion) and/or Section 2c. Unless you really know what you're doing with regards to software licensing, I would highly suggest not modifying the welcome message. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk 12 - CDR changes
On Wed, Mar 19, 2014 at 5:13 AM, jg wrote: > Can you publish a short stub of your dial plan to see what you are doing? > There are the NoCDR, ForkCDR, and ResetCDR applications that might help. > > jg > Due to core changes that occurred in Asterisk 12, CDRs were completely redone. There wasn't much way around having to do that; the old CDR behaviour was tightly integrated with the old bridging code, and when that bridging code went, CDRs had to be get reworked as well. This was noted strongly in both the UPGRADE notice delivered with Asterisk, as well as on the Upgrading to Asterisk 12 wiki page: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12 Depending on your dialplan - and what the channel does as it moves through the dialplan - it would not be surprising to find records you didn't have in the past. The philosophy with CDRs in Asterisk 12 is to show you what the channel did as it moved around, rather than try to hide information. In the past, Asterisk would frequently get this 'hiding' wrong, much to many people's consternation. This does mean you may have to post-process your records; luckily, as jg pointed out, there are still plenty of ways to manipulate CDRs through the dialplan. A specification for CDR behaviour in Asterisk 12 is available on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Wrong patch 1.8.26.1 at http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.26.1-patch.gz ?
On Sun, Mar 16, 2014 at 5:55 AM, Administrator TOOTAI wrote: > Hi, > > I have patch failure trying to apply asterisk-1.8.26.1-patch.gz I took a > look in the patch and found those lines at the begining: > > --- asterisk-1.8.16.0-summary.txt (.../1.8.16.0) (revision 410440) > +++ asterisk-1.8.16.0-summary.txt (.../1.8.26.1) (revision 410440) > @@ -1,221 +0,0 @@ > -Release Summary > - > - asterisk-1.8.16.0 > - > -Date: 2012-09-13 > - > - > > This doesn't seems OK to me ... > > -- > Daniel Hi Daniel: Thanks for catching that. I took a look at the release and the tarball itself was made properly; however, the patch file was generated incorrectly. The patch did still apply the security fix, but since it was generated against a much older version, it would have been difficult to apply to 1.8.26.0. I've updated the patch on the downloads site such that it is now a patch against 1.8.26.0. Let me know if you have any other issues. Thanks - Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Any help Address 0xfffffffe out of bounds in app_confbridge.casterisk-11.5.1 using confbridge.conf
On Thu, Mar 13, 2014 at 9:24 AM, Rusty Newton wrote:
> On Thu, Mar 13, 2014 at 4:18 AM, hkc323 wrote:
>> Address 0xfffe out of bounds why and how to
>> solve.MyConfbridgeCount(conferencenumber,variablename )return total number
>> of user in conference given by conferencenumber otherwise zero.At runtime
>> using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call
>> function count_exec(struct ast_channel *chan, const char *data).But at
>> compile time char * data cause core dumped. Asterisk-11.5.1 Centos6
>> app_confbrige.c confbridge.conf
>> =
>> Task: Using Dailplan user want to retrive no of user in conference
>> '6050' => 1. Verbose(3,"testMyConfbridgeCount") [pbx_config] 2.
>> MyConfbridgeCount(4000,count) [pbx_config] 3. verbose(3,"== ${count} ")
>> [pbx_config]
>
> Please discontinue spamming the users list with your posts.
>
> Not receiving an answer to your question is not a reason to repeatedly
> post (four posts now in the past few days?)
>
> I've already responded to your original post and asking you to post on
> the issue tracker and follow the issue guidelines to provide the
> information needed to investigate the crash.
>
Actually, in this case, he shouldn't post a bug report to the issue
tracker. The bug he is encountering is with some custom code in
ConfBridge, namely with the application "MyConfbridgeCount":
static const char *const app2 ="MyConfbridgeCount";
You should contact the author of that code and ask them to fix the crash.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
On Tue, Mar 11, 2014 at 11:23 AM, Yaron Nachum wrote: > Hi Mathew, > The regular sip stack has 'auto' dtmfmode which behaved as I said - if the > remote replied with telephony event it used RFC2833 otherwise it used > inband. > Correct. There is no setting for dtmf_mode that is analogous to the chan_sip 'auto' setting - what you configure for you endpoint today is what it will use. That's not a bug, just something not existing yet. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
On Tue, Mar 11, 2014 at 8:23 AM, Yaron Nachum wrote: > Hello, > I have installed the latest version 12 that has been released (12.1.0.rc3). > > I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint > that doesn't support it (no telephony event in the rtpmap) the asterisk > responds OK in the signalling but DTMF is not working. > > Is it a known issue? > I don't think that's an issue at all. Your configured your endpoint to support RFC 4733 DTMF. However, the INVITE request that was received by Asterisk didn't offer support for DTMF, so Asterisk can't accept it. It has to accept only what is in the offer. Your configuration can't force the UA to offer what it wants - you can only configure Asterisk with what it should support with that UA. There's really only two possible outcomes here: (1) Reject the INVITE request with a 488 (you didn't offer me DTMF!) (2) Accept the INVITE request but not have DTMF over RFC 4733. What you're seeing is option (2), which I think is better than rejecting the entire call simply because the thing you are talking to doesn't support the DTMF mode you configured it to have. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] PJSIP - Using multiple AOR contacts when dialing through an endpoint
On Tue, Mar 11, 2014 at 8:45 AM, Yaron Nachum wrote: > > Thanks for the response anyway. > > I think that it would be great if someone would make it happen. It seems to > me trivial that once you enable to setup multiple AORs you would use them :-) > > Yaron. > > > On Tue, Mar 11, 2014 at 3:38 PM, Joshua Colp wrote: >> >> Yaron Nachum wrote: >>> >>> Thanks Joshua, >>> I tried it already. That would generate a call to both AORs which is not >>> what I was looking for. >>> >>> Isn't there a way to retrieve the AOR status from the dialplan? >> >> >> Not currently. >> We're still adding dialplan functions and CLI commands to the PJSIP stack. Right now there's a way to drill down into endpoint configuration via the PJSIP_ENDPOINT function, but we haven't yet expanded that to AORs. Doing so is a pretty natural next step. There's some discussion of this on the following JIRA issue, where Josh mentions we could query down into the contacts for some of the information: https://issues.asterisk.org/jira/browse/ASTERISK-23173 We'd probably have something similar to PJSIP_ENDPOINT, such as PJSIP_AOR or PJSIP_CONTACT (or something like that), that lets you get at the run-time information of an AOR. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Add SIPCALLID of egress leg to CDR
On Mon, Feb 24, 2014 at 1:38 PM, Ricky Hartmann wrote: > Hey all, > > I've been fighting with this all morning, and I feel like this should be a > relatively simple task, but I just can't get it to work. I currently have a > very basic asterisk v11.6 setup with a single extension (a Bria softphone) > and a single sip trunk to my carrier. > > What I'm trying to accomplish is simply adding the asterisk generated > SIPCALLID of the leg between asterisk and the carrier to the CDR when it's > written. (This is in addition to the SIPCALLID of the original call leg, > which I've already added.) I've been able to print the SIPCALLID for the > carrier leg to the CLI if I jump into a Macro or Gosub from the Dial command > (using the M or b options respectively), but that's about it. > > From what I can tell, this should be easily accomplished with variable > inheritance, but based on what I'm seeing during test that seems to be a one > way relationship, as updates in a child channel don't seem to reach the > parent. Am I missing something? Is there a better way to go about this? > > The few questions I've found on forums and mailing list archives on this > subject are all from 2007 and earlier, so I'm hoping something new has come > along in the last few years and I just haven't found it anywhere. Can > anyone help? > You're correct - variable inheritance only goes in one direction, from the parent to the child. Since the child channel is going to be Party B in the CDR, variables set on it are unlikely to show up. I can think of a few options: 1) Use the userfield on the outbound channel to store the SIPCALLID. A little known fact of the userfield is that, on two channels in a bridge together, the userfields are concatenated together using a ';' as a delimiter. 2) Use the MASTER_CHANNEL function to reach back to the parent channel and set the CDR variable there. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] h extension isn't processed after call file finishes.
On Tue, Feb 18, 2014 at 2:13 PM, Steve Edwards wrote: > On Mon, 17 Feb 2014, Mike Diehl wrote: > >> Is there something I need to do in order to get the h extension to get >> called? > > > Would the 'g' dial() option help? > > "Proceed with dialplan execution at the current extension if the destination > channel hangs up." > > It won't take you to h, but it may allow you to do what you need to do -- > even if the next dialplan priority just says 'goto h.' > I'm actually a bit confused about what channel(s) are executing the 'h' extension. From the description in OP's e-mail, it sounds as if at least one channel is dropping into the 'h' extension, and some channels are not. Which channels are they? If it is the outbound channel, then since that channel doesn't execute dialplan, it will never get put into the 'h' extension, unless you use the Dial application's 'e' option. If you want hangup logic and you're using Asterisk 11+, you could also use a hangup handler on the outbound channel. But otherwise, I would expect that the 'h' extension would always be fired for a channel executing dialplan, so long as it is in the same context. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Retaining P-Asserted Info
On Mon, Feb 17, 2014 at 4:29 AM, Nick Cameo wrote: > Hello Ishfaq, > > I just tried it and it did create a P-Asserted header however it > contains the extension > of the asterisk peer not what was passed by our switch. From the > previous example: > > P-Asserted-Identity: "222" (which is asterisk > peer extension and not) > P-Asserted-Identity: "John Doe" > ; user=phone; nat=yes. (which is > being passed by the call leg) > > Is there a flag that retains the rpid from the call leg? > No. Asterisk is a back to back user agent, not a proxy. Overriding the settings of a peer with the peer that it is bridged with is typically contrary to Asterisk's nature. If you want to copy information from one SIP channel to another, you should do as Markus suggested. Matt -- _ -- 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] how to get full channel name - AMI cuts off [solved]
On Thu, Jan 30, 2014 at 5:48 PM, Justin Killen wrote: > After posting this, I ran across 'core channel show concise', which gives > the data in a more machine friendly format. > > That may work over AMI, but in general, it isn't recommended. The command class authorization, EVENT_CLASS_COMMAND, is relatively powerful and shouldn't be exposed to a general AMI action without a lot of forethought. What's more, CLI commands are generally viewed as being appropriate for end users, and not programs controlling Asterisk. While 'core show channels concise' is unlikely to change in future versions, it certainly isn't versioned in the same fashion as AMI events/actions. If you need to get a dump of all active channels in the system over AMI, I'd recommend the CoreShowChannels AMI action [1]. It will send the information back for each channel as an event, and doesn't require the same level of permission as the corresponding CLI command. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_CoreShowChannels Matt -- _ -- 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] Parking in Asterisk 12.0.0
On Thu, Jan 30, 2014 at 2:58 PM, Leandro Dardini wrote: > I have converted the normal Park application and I can only alert you about > the syntax change. I suspect also in the ParkAndAnnounce command, the > parameters are ordered completely different. > > Leandro > > Please go ahead an open an issue for this - issues.asterisk.org. The problem here is that you are attempting to enter into a Parking bridge while you are still technically in a bridge. The DTMF features that account for the 'normal' mechanism of doing this - the one touch parking feature - recognize that you are in a bridge and do a safe transfer from the existing bridge to the parking bridge. By jumping out to a macro/gosub and directly going in through the ParkAndAnnounce application, you are bypassing that logic. The code in bridge_channel_internal_join is preventing you from going into the parking bridge as it knows that you have not yet safely left the bridge you are in. We'll take a look and see if there's a way to allow this to happen again. For now, you should use the one touch parking feature. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] [HELP]: Auto-answering calls placed from call files
On Tue, Jan 28, 2014 at 10:56 AM, Steve McCann wrote: > Hello All, > > I've asked this on the asterisk-dev list, so sorry for cross-posting. So far > I'm not sure how to accomplish this without looking at the source code or > looking at some other way to get around this issue. > > > I'm trying to have an automated call to an Aastra SIP phone and have the > call auto-answeredby the phone. I know that a SIP call placed to the phone > can be auto-answered if a certain SIP header is added to the call. I am able > to apply the SIP headers manually and get that working (using > SIPAddHeader(Alert-Info: info=alert-autoanswer) in the dialplan, but for > call files, I don't seem to be able to edit any of the sip headers - there > is only basic customizations allowed to setup the calls. > > Does anyone know how I could place automated outgoing calls that would have > the proper sip headers added to it that would allow the call to be > auto-answered? > > I've also posted this question to the forums here: > http://forums.asterisk.org/viewtopic.php?f=1&t=89190 > > Many thanks, > Steve > This isn't a development question, as it doesn't relate to the actual Asterisk source code itself. Cross-posting across the -dev and -users lists isn't helpful either, as pretty much everyone who is subscribed to the asterisk-dev list is also subscribed to the asterisk-users list. As SIPAddHeader is a dialplan application and not a dialplan function, it cannot be used from a call file. One approach to performing an outbound call that requires SIPAddHeader - and that doesn't rely on undocumented behaviour - is to use the call file to create a Local channel in the dialplan that dials the SIP channel, and use SIPAddHeader from there. A quick Google indicates others have used a similar approach in the past as well [1]. [1] http://lists.digium.com/pipermail/asterisk-users/2008-January/204375.html Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] grp_lock error when compiling against pjproject
On Tue, Jan 28, 2014 at 2:40 AM, Ira wrote: > Hello Matthew, > > > Monday, January 27, 2014, 1:49:44 PM, you wrote: > > > Do you have the exact error message that pjproject gave when you ran > into this problem? > > I don't, but I guess I can reinstall the offending software to get it if > you need it. It's documented on the bug list as I eventually found the > using google. > I'm including the asterisk-users mailing list on this reply, as there's no reason to take this discussion off list. I'm not sure what bug list you're referring to. However, the page on the wiki that documents common errors and their appropriate correction [1] attempts to provide the exact error message that users will see when they encounter that situation. I'd be happy to update it with whatever error you ran into, but to do so we need to know the exact messages. Alluding to error messages without providing them usually leads to more confusion, not less. [1] https://wiki.asterisk.org/wiki/display/AST/Installing+pjproject#Installingpjproject-IssuesandWorkarounds Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] IOPS required by Asterisk for Call Recording
On Mon, Jan 27, 2014 at 1:02 PM, Ron Wheeler wrote: > Can you get a reading of the total number of I/Os during your test? Peak > IOPS? > That might tell you very quickly about the storage pattern that Asterisk > uses. > > Can you configure a RAM drive to see if disk is really the bottleneck. May > need to add some more RAM memory to your configuration. > > What is your network capacity? Usually one can write faster than the network > can deliver - just to make sure that you are chasing the right bottleneck. > > What happens at 80 calls to tell you that you have run out of IOPS? Dovetailing on this question, I'll add one as well: Are you recording using MixMonitor, or Monitor? Depending on your answer to the "what happens at 80 calls", you may get better results with MixMonitor over Monitor. MixMonitor offloads the recording of the media to a separate thread; Monitor attempts to record the audio on the thread servicing the channel(s). Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] grp_lock error when compiling against pjproject
On Sat, Jan 25, 2014 at 1:35 PM, Ira wrote: > Hello Asterisk, > > Would someone be kind enough as to add the issue: > > grp_lock error when compiling against pjproject > > and solution: > > delete the rogue install in /usr/local/include > > To the WIKI page about installing pjsip. > > I tried to update the WIKI but don't seem to have a way to do it. > > I know it's not supposed to happen and I know what I did wrong, but it's > hard to imagine I'll be the last person to make that mistake. > > > Do you have the exact error message that pjproject gave when you ran into this problem? -- Matthew Jordan Digium, Inc. | Engineering Manager 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] AMI eventmask question
On Thu, Jan 23, 2014 at 9:31 AM, Michelle Dupuis wrote: > Thanks - I've been through that doc before and couldn't find the info > needed, which is why I went to the source code eventually. > > All events are grouped, and each group is given a name/flag like 'system', > 'call', etc. The docs just don't say which events are in which group/flag. > > Perhaps something Digium could add at some point :) Or someone from the open source community... this is an open source project, after all :-) The managerEventInstance XML elements already have an attribute for the manager class, which is populated: Raised when a parked call times out. The parking lot extension. The name of the parking lot. ParkedCall You could actually grep the core-en_US.xml file and get all of the events that match to a particular class authorization. It doesn't show up in the CLI due to the xmldoc API not parsing out that attribute. The same is true for the wiki documentation; that project is up on github [1]. It wouldn't be a large patch to either to have that attribute displayed. Matt [1] https://github.com/asterisk/publish-docs -- Matthew Jordan Digium, Inc. | Engineering Manager 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] CDR and Transfer, an asterisk scaring bug lasting from 1.4 version...
On Thu, Jan 23, 2014 at 12:44 PM, Leandro Dardini wrote: > When you use a product which version number is 11 or even 12, you might go > with the assumption all big bugs are fixed and then you find there is a > huge, important, expensive bug still running in the code we are relaying > upon... First, not all versions in 11 are the same. Bugs do get fixed. What version of Asterisk 11 are you using? Second, CDRs are not the same in Asterisk 12. Due to extensive changes in the bridging core, CDRs were re-worked heavily. You may want to take a look at the notes on the Asterisk wiki [1] for Asterisk 12, as well as the CDR specification for Asterisk 12 [2]. > The problem is simple. If you transfer a call, that dialing will be not > reported in the CDR, so no billing will happen. This is a simple example: And how did you do the transfer? Via DTMF features? Via a particular channel driver technology? If so, which channel drivers were involved? What kind of transfer was it? Blind? Attended? Failed attended (the notorious blonde transfer)? > > Extension 100 calls extension 101 > After 10 seconds, extension 100 transfer the call to > 00VERYEXPENSIVEDESTINATION > After 100 seconds, extension 101 hangup the call > > What do you find in the CDR? Just one record for a call from extension 100 > to extension 101 lasting 10 seconds. What about the 100 seconds call from > 100 to 00VERYEXPENSIVEDESTINATION? It will never get billed. > > How do you manage these cases? > I'm not sure if there is a bug report filed against CDRs for the currently maintained branches for lost records during a blind or attended transfer that matches your issue. There is ASTERISK-17826, which may or may not be your issue: the noted lack of information makes it a bit hard to tell. The last issue that I'm aware of that we fixed regarding lost CDRs during a transfer was ASTERISK-21394, which was fixed in 11.4.0. So, if you're using a version prior to 11.4.0, you may want to consider upgrading. Again, due to the lack of information, it's hard to tell whether or not that would help you. Finally, CDRs in versions of Asterisk prior to 12 are subject to the whims of channel masquerades. This has historically made it difficult, if not impossible, to guarantee correctness during all transfer operations. Additionally, even if we could guarantee a particular set of behaviour in all circumstances, the lack of any clear agreement as to what a CDR should look like after an attended transfer (or in any situation that involved multiple parties) made the problem impossible to solve to the satisfaction of everyone. This particular reason is why CEL was created. If you continue to have problems with the billing records, you may want to consider moving your billing logic to CEL. Note that since (a) Asterisk 12 re-architected using a consistent bridging framework, which killed visible channel masquerades; and (b) we decided to not try and please everyone and just defined CDRs for how we thought they should work; the behaviour of CDRs in Asterisk 12 and in future versions should be substantially more predictable. Matt [1] https://wiki.asterisk.org/wiki/display/AST/New+in+12 [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk Community Code of Conduct
community resources including the bug tracker, mailing lists, wiki, version control, continuous integration services, and other necessary project infrastructure. Asterisk goals and objectives are decided upon along with the community at the annual AstriDevCon held at AstriCon. Development discussions take place on the public asterisk-dev mailing list and the #asterisk-dev IRC channel. More information on the development of Asterisk can be found on the Asterisk wiki. [1] https://wiki.asterisk.org/wiki/x/0gifAQ The Asterisk project would not be what it is without the community, and the continued success of the Asterisk project is directly dependent on the success of the community. As always, thank you all for your continued support of the Asterisk project - and the Asterisk community! -- Matthew Jordan Digium, Inc. | Engineering Manager 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] asterisk 11.7.0: Delayed audio
On Fri, Jan 10, 2014 at 9:45 AM, gm1 wrote: > On connection to an incoming call via PSTN where > asterisk [11.7.0] is Dialing an internal extension > on answering the call there is about 6-7 seconds before > audio is heard on either side. > > > When looking at the CLI traces when I answer the incoming call that asterisk > extensions were dialing, I see immediately upon answering >>0xhexnumber -- Probation passed - setting RTP source address to >> 192.168.1.11:portnumber > then not until about 6 seconds later I see this >>0xhexnumber -- Probation passed - setting RTP source address to >> 192.168.1.11:diffportnumber > and immediately hear audio > > what appears to be an issue is that the RTP link(audio) setup is delayed. > > > Anyone have suggestions on how to fix this issue? > If the RTP source address/port is changing, then Asterisk is receiving RTP packets from two different sources and is waiting for one of them to stabilize before it picks the actual source of the media stream. That's by design, as the "locking in" of the RTP source prevents a media injection attack. You can tweak how Asterisk does this using two settings in rtp.conf: ; Enable strict RTP protection. This will drop RTP packets that ; do not come from the source of the RTP stream. This option is ; enabled by default. ; strictrtp=yes ; Number of packets containing consecutive sequence values needed ; to change the RTP source socket address. This option only comes ; into play while using strictrtp=yes. Consider changing this value ; if rtp packets are dropped from one or both ends after a call is ; connected. This option is set to 4 by default. ; probation=8 Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Failed to get 160 samples from read factory , asterisk-11.5.1 app_confbridge.c
On Thu, Jan 9, 2014 at 2:59 AM, hardik wrote: > http://stackoverflow.com/questions/21015596/failed-to-get-160-samples-from-read-factory- > asterisk-11-5-1-app-confbridge-c > 1. Please don't post a link to another site with no explanation or summary of your question. Not only are you unlikely to get many responses with this approach, but some would consider it a bit rude. 2. As your post on StackOverflow is pertaining to Asterisk development, your question (whatever it happens to be) belongs on the asterisk-dev mailing list [1]. 3. This looks similar to your previous ConfBridge/DTMF question on the asterisk-dev mailing list [2], although since you haven't replied there, I'm not entirely sure. Either way, the advice in my reply still stands: please take the time to provide context to your question and to the code that you are asking about. Sending a large block of code that you've written with no context and asking someone to debug the problems you're seeing is unlikely to generate the help you want. Thanks - Matt [1] http://lists.digium.com/mailman/listinfo/asterisk-dev [2] http://lists.digium.com/pipermail/asterisk-dev/2014-January/064504.html -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk 10 EOL Notice
Hello everyone! On December 15th, 2013, Asterisk 10 officially reached its End of Life [1]. As a Standard Release, Asterisk 10 received one year of bug fix support, followed by one year of security fix support. Users of Asterisk 10 should consider moving to Asterisk 11 at their earliest possible convenience. Asterisk 11 is a Long Term Support (LTS) Release, and will continue to receive bug fix support until October 25th, 2016, with security fixes continuing through October 25th, 2017. We'd like to thank everyone who contributed bug reports, test reports, patches, and features to Asterisk 10. Asterisk 10 contained many new features (ConfBridge! T.38 Gateway! Core support for wideband audio! Media format attribute negotiation!) that helped significantly advance the Asterisk project. Asterisk 11 has been a successful LTS release in large part due to the hard work that was done in Asterisk 10. So, to everyone who helped make Asterisk 10 successful, thank you! Matt [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions -- Matthew Jordan Digium, Inc. | Engineering Manager 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] A Question about Management/Control Protocol Licensing
On Wed, Dec 11, 2013 at 3:15 PM, Paul Belanger wrote: > On 13-12-11 03:15 PM, Steve Murphy wrote: > >> I see the following paragraph in the Asterisk trunk LICENSE file: >> >> "In addition, Asterisk implements two management/control protocols: the >> Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface >> (AGI). It is our belief that applications using these protocols to >> manage or control an Asterisk instance do not have to be licensed >> under the GPL or a compatible license, as we believe these protocols >> do not create a 'derivative work' as referred to in the GPL. However, >> should any court or other judiciary body find that these protocols do >> fall under the terms of the GPL, then we hereby grant you a license to >> use these protocols in combination with Asterisk in external >> applications licensed under any license you wish." >> >> This probably originated some years ago, and I wonder if Digium or the >> Asterisk >> community might consider adding the OTHER management/control protocols to >> this >> list: ARI, and the ExternalIVR interface. >> >> If not, it might be instructive to learn why! >> >> Would also like to see this update to include ARI. We talked a little > about it at astridevcon, and I think it is likely an oversight. > > It isn't an oversight. It's on my ToDo list (and this item is an action item on the wiki as well). We had the Thanksgiving holiday; then I was out last week at AdhearsionConf (great conference!). The licensing file will get updated before 12 is released. As an aside, we also had conversations about it on the asterisk-app-dev list [1], where I responded that I would get answers to the licensing questions. Granted, it has been much longer than a week or two - mea culpa on a bad time estimate. [1] http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000127.html Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Not able to get remote channel variables containing RTCP values
On Mon, Dec 2, 2013 at 10:54 AM, Thomas Rechberger
wrote:
> I am not sure if its just me, but i am able to get only local channel
> variables containing RTCP QOS values.
> The Version is 1.8.14.
> I want to store values of bridged channel in CDR.
>
> Phone is Cisco 7941 SIP and with sip show channelstats i see all the
> relevant information (jitter,packet loss) i want to get. It even calculates
> packet loss in %. But i am not able to store it to CDR.
>
> Asterisk 1.4 seems to have had a function ast_rtp_get_quality but i cant
> find any information about that in sources from 1.8, only a short reference
> in 1.4.
>
> Channel variables like CHANNEL(rtpqos,audio,rxjitter) show only
> information about the local channel. So not really usefull.
> In some old version they seemed to have it changed from remote_jitter to
> rxjitter, local_jitter to txjitter and so on. Was not even documented.
>
The values that can be extracted are documented:
https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL
I'll admit the description of each parameter isn't terribly verbose, but
there is documentation for what they are. The local/remote refers to the
values reported by Asterisk (local) and the endpoint it is communicating
with (remote). This is only for a single channel; bridged channels are not
accessed by this function.
> The 2 variables RTPAUDIOQOSBRIDGED and RTPAUDIOQOS show exactly the things
> i want, but all information is stored in one field so its not really usable
> because it looks ugly in CDR report and doesnt show packet loss in %.
>
If you wanted, you could parse the values in those channel variables. They
are semi-colon delineated lists with fixed fields, so if there are
particular values you want you can extract them. This does mean doing a bit
of string parsing in the dialplan, but it is a viable option.
>
> The following interesting variables are completely empty (show 0), here is
> how i write it to CDR in h:
> exten => s,n,Set(CDR(txj)=${RTPAUDIOQOSJITTER})
> exten => s,n,Set(CDR(rxj)=${RTPAUDIOQOSJITTERBRIDGED})
> exten => s,n,Set(CDR(txpl)=${RTPAUDIOQOSLOSS})
> exten => s,n,Set(CDR(rxpl)=${RTPAUDIOQOSLOSSBRIDGED})
> exten => s,n,Set(CDR(txrtt)=${RTPAUDIOQOSRTT})
> exten => s,n,Set(CDR(rxrtt)=${RTPAUDIOQOSRTTBRIDGED})
>
> I also checked variables during call with featurecode, but also empty.
>
> Did i oversee something? Is it the same in Version 11 ?
> I dont want to mess with Voipmonitor because i only need 2 variables of
> remote channel. If sip show channelstats is showing everything correctly,
> it should be not that hard to get that information.
Since you want to write this into the inbound channel's CDR, there isn't
much of another option other than to parse out the channel variables, even
in Asterisk 11.
Technically, in 11, you can use a Pre-Dial handler to attach a Hangup
Handler to the outbound channel. The hangup handler is a subroutine that
will be executed on the outbound channel when it is hung up. In the hangup
handler, you can use the CHANNEL function to extract the values directly
from the outbound channel - however, you can't use that to modify the CDR
on the inbound channel, so that's of limited use to how you want to use it.
Your best option would be to parse out the values in the various channel
variables and store the ones you want.
As an aside: as time has gone on, the idea of always reaching across a
bridge to get another channel's values has become less favoured. Such a
concept doesn't do well with ad-hoc multi-party bridges or conferences, and
thus isn't always sustainable in all scenarios. For the most part, the
emphasis in latter versions is to give people access to the specific
channel that they want to manipulate/retrieve information from, as opposed
to relying on the two-party nature of bridges.
This usually works pretty well, except for CDRs, which are generally a mess
no matter what. :-)
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] Voicemail greeting playback issues?
On Mon, Nov 25, 2013 at 7:17 PM, Patrick Lists < asterisk-l...@puzzled.xs4all.nl> wrote: > On 11/26/2013 12:24 AM, Doug Lytle wrote: > > Bryant Zimmerman wrote: > >> Hey all > >> > >> I believe I found the bug in Asterisk 11.xxx If someone can help me > >> verify it. > > > > Actually, > > > > I wouldn't consider it a bug. I've know for years that you need to > > answer a channel before you play back audio or strange things can and > > will happen. > > That's what I do since the 0.x days. IIRC in recent Asterisk versions > some apps answer before doing anything else. Guess the voicemail app is > not one of them. I always answer first followed by a small Wait and then > execute the app. > > VoiceMail does automatically Answer a channel. I'm going to guess that you have strictrtp enabled (which it is by default), and that if you cranked up Asterisk verbose logging to at least 4, you'd see something like this at about the time you started hearing audio: 0xYY - Probation passed - setting RTP source address to xxx.xxx.xxx.xxx Asterisk drops RTP packets until it locks onto an RTP source. It does this to prevent media injection attacks. The default probation period for an RTP source is four packets - you can configure the probationary period as well as whether or not strict RTP checking is enabled in rtp.conf. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk 10 EOL Approaching
Hello! This is a friendly reminder that the official End of Life of Asterisk 10 is approaching soon. As a Standard release of Asterisk, Asterisk 10 received one year of maintenance fixes followed by one year of security fixes; that final year is now just about up. After 2013-12-15 (December 15th), releases of Asterisk 10 will no longer be made. Users of Asterisk 10 are encouraged to move to the next major version, Asterisk 11, as soon as possible. Asterisk 11 is a Long Term Support (LTS) and has maintenance support through 2016-10-25, with its full End of Life occurring on 2017-10-25. For more information on Asterisk versions and their supported lifetimes, please see the following wiki page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Thank you for your continued support of Asterisk! -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk 1.8.20 crashing
On Tue, Nov 12, 2013 at 7:56 AM, Amit Patkar | ATPL wrote: > > Hi > > I am experiencing Asterisk Crash. Log got stopped when asterisk crashed. > Please help me to identify the reason and fix this issue. > > Asterisk: 1.8.20 > > I am using AMI and fastAGI to control the call. Some part of dial plan is > also defined in extensions.conf > > I am experiencing this crash when app_meetme conference functionality is > used with more than 3 parties. I faced this issue with app_conference also. > This is finally resulting in call drops. > Please help me to identify and resolve this issue. > Last few lines of log are copied in this email. I can email full log, if > this does not give sufficient information of crash. > > While the log will be helpful, I'd recommend getting a backtrace from the core produced when Asterisk crashed. Instructions on generating a proper backtrace can be found on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace Please make sure Asterisk and its modules are compiled with DONT_OPTIMIZE and BETTER_BACKTRACES. When you have a backtrace, please file an issue in the Asterisk issue tracker - attach the log as well, as it may be useful in analyzing how the system got into that state. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] [asterisk-dev] how determine mandatory modules to slimming asterisk
On Mon, Nov 11, 2013 at 1:10 AM, s m wrote: > hello guys > > i want to slimming my asterisk by loading only mandatory modules. in order > to do that, i edit my modules.conf file and set autoload=no and load just > mandatory modules. > > my problem is, how should i determine which modules are necessary to > asterisk works correctly? i have sip, h323 and dahdi connection on my > asterisk. is there any documentation about mandatory modules for asterisk? > or anybody has such a list? > > any comments or hints are appreciated > SAM > First, please don't cross post across mailing lists. The asterisk-users mailing list is an appropriate place for questions about Asterisk deployment; the asterisk-dev mailing list is appropriate for questions regarding development, i.e., code. As many people subscribe to both lists, cross posting is not necessary. Since this is a question about deployment, it is appropriate for the asterisk-users mailing list, and not the asterisk-dev list. Second, there is not a canonical list of 'required' modules for Asterisk. You can run Asterisk without any of the loadable modules and it will technically function, although it obviously won't do much. What modules you need is determined by your deployment - which includes both the channel technologies you want to use, as well as your dialplan. That isn't something anyone can answer for you. You can, however, use menuselect to help determine what module dependencies need to be fulfilled. Beyond that, you should look at what applications and functions a module provides to determine if you need it. Asterisk: The Definitive Guide has some excellent information in Chapter 2. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] warnign
On Thu, Oct 24, 2013 at 11:41 AM, troxlinux wrote: > Hi, make the change in the file chan_h323.c and comment these lines > > > // default: > // ast_log(LOG_WARNING, "OH323: Don't know how to indicate > condition %d on %s\n", condition, token); > // break; > > but the message keeps coming, I'm doing something wrong? > > WARNING[19460][C-]: chan_ooh323.c:1413 ooh323_indicate: Don't > know how to indicate condition 33 > > > File a bug in the Asterisk issue tracker: https://issues.asterisk.org/jira Make sure you mark the component as chan_ooh323. Alexandr Anikin is the maintainer of chan_ooh323 [1]; he may be able to correct the issue for you. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Open+Source+Maintainers -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk AMI 1.3 Specification
On Wed, Oct 23, 2013 at 6:17 PM, Shishir Pokharel wrote: > Thanks Rusty. > Do you happen to know is there any changes on data of response Key for > Manager Actions which exists in both AMI 1.0 and AMI 1.3 ? > > > There are most likely many such changes between 1.0 and 1.3. You'll need to look at the various CHANGES files for each version to determine what changed in AMI between 1.0 and 1.3. Starting on the Asterisk wiki with the various "New in X" pages is also a good idea: * https://wiki.asterisk.org/wiki/display/AST/New+in+1.8#Newin1.8-AsteriskManagerInterface * https://wiki.asterisk.org/wiki/display/AST/New+in+10#Newin10-AsteriskManagerInterface * https://wiki.asterisk.org/wiki/display/AST/New+in+11 -- Matthew Jordan Digium, Inc. | Engineering Manager 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] warnign
On Wed, Oct 23, 2013 at 7:12 PM, Vladimir Mikhelson wrote: > Richard, > > And what is "condition 33" after all? Maybe it needs to be processed, not > ignored. > > The various indication control frames are defined in frames.h. Frame type 33 was added to Asterisk 11: AST_CONTROL_PVT_CAUSE_CODE = 33, /*!< Contains an update to the protocol-specific cause-code stored for branching dials */ This is part of the hangup cause feature that was added to Asterisk 11. When a channel driver raises this control frame, the channel core uses the information in the frame to update protocol specific cause code information. Channel drivers are supposed to indicate to the core that they don't want to handle that frame, i.e., return -1. So Richard is correct: chan_ooh323 should, in ooh323_indicate, handle the frame by returning -1. Actually raising the frame to pass a channel technology specific hangup cause code is completely up to the channel driver. Not all channel drivers support it; if someone wanted to add that functionality to chan_ooh323 that'd be great; but it's completely different than the condition that the OP is seeing. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk-12 issue after successful installation
On Mon, Oct 21, 2013 at 7:59 AM, A J Stiles
wrote:
> On Monday 21 October 2013, virendra bhati wrote:
> > Hi Team,
> >
> > I have installed asterisk-12 Beta but when I try to asterisk start then
> get
> > below issue.
> >
> > *[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r
> > asterisk: error while loading shared libraries: libjansson.so.4: cannot
> > open shared object file: No such file or directory
> > [root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]#*
>
> Did you build it yourself from Source Code, or did you install someone
> else's
> pre-compiled package? If the latter, the packager may have omitted a
> dependency. It happens from time to time. You probably need to install a
> -
> dev or -devel package (what distro are you running?)
>
> What do you get for
> # ldd /usr/sbin/asterisk
> ?
>
Hello -
libjansson is now a required library. Please see the build system changes
in the UPGRADE notes [1] or on the wiki [2]. Note: if you have not yet read
the upgrade notes and the list of changes, please do so before installing
and running Asterisk 12. Please :-)
Note that if your distro doesn't have a package of libjansson (or, more
accurately, libjansson-dev{el}), you can download a source tarball and
install it [3]. The install_prereq script [4] should also take care of it
for you.
[1] http://svn.asterisk.org/svn/asterisk/branches/12/UPGRADE.txt
[2] https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12
[3] http://www.digip.org/jansson/
[4]
http://svn.asterisk.org/svn/asterisk/branches/12/contrib/scripts/install_prereq
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] Launched new Asterisk Cilck 2 Call for websites
On Tue, Oct 15, 2013 at 7:31 AM, Vignesh Prajapati < vignesh.prajap...@enjayworld.com> wrote: > Want VOIP Click 2 Call solution? We have designed a tool for providing the > same with the help of Google chrome extension by integration of Asterisk > and Google Chrome APIS. > > Try Google chrome extension with your Asterisk, believe this will reduce > your call efforts. > > Visit here - > http://www.ens.net.in/index.php/products/synapse/synapse-products-add-ons/synapse-browser-extension<http://www.ens.net.in/index.php/products/synapse/synapse-products-add-ons/synapse-browser-extension?utm_source=chrome_extension&utm_medium=click_help&utm_campaign=synapse> > for > more information. > > If you find any difficulties at time of configuration then let us know, we > will help you. > > Regards, > -- > Vignesh Prajapati > [image: Inline image 10] +91 0260 6612911 > [image: Inline image 8] <http://www.ens.net.in/> > > [image: Inline image > 1]<http://www.linkedin.com/company/enjay-it-solutions-ltd-> > [image: Inline image 2] <http://www.facebook.com/enjayltd> [image: > Inline image 3] <https://twitter.com/enjayworld> [image: Inline image > 5]<https://plus.google.com/104619851885354132849/posts> > [image: Inline image 2] <http://eitsl.blogspot.in/> [image: Inline image > 7] > > This is not an appropriate message for the asterisk-users mailing list. The asterisk-users list should be used for discussions regarding Asterisk deployment and usage. You've already sent this message to the asterisk-biz list, which is the appropriate forum for commercial discussion. Please do not post messages like this again to the asterisk-users list. > *Disclaimer:* > This message contains privileged and confidential information and is > intended only for an individual(s) named. If you are not the intended > recipient, you should not disseminate, distribute, store, print, copy or > deliver this message. Please notify the sender immediately by e-mail if you > feel that you have received this e-mail in error and you are advised to > please delete this e-mail from your system. E-mail transmission cannot be > guaranteed to be secure or error-free as information may be intercepted, > corrupted, lost, destroyed, arrive late, be incomplete or contain viruses. > The sender does not accept any liability for any errors or omissions in the > contents of this message which arise as a result of e-mail transmission. If > verification is required, please request a hard-copy version. > > Please don't print this e-mail unless you really need to. > FYI: This is a silly disclaimer when sent to a public mailing list. (And yes, I don't feel I should have received this.) Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] New mailing list - asterisk-app-dev
Hey all - After much discussion at AstriCon, it became clear that the Asterisk project could use a mailing list dedicated specifically to application development. This new mailing list should be used specifically for discussions regarding the development of applications using AMI, AGI, or ARI - or any other interface exposed by Asterisk in the future. Today, we're pleased to announce the asterisk-app-dev mailing list, now available on lists.digium.com: http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev This new list is not a replacement for asterisk-dev or asterisk-users. The asterisk-dev list should continue to be used for discussions regarding development directly in the Asterisk project itself or the Asterisk Test Suite. The asterisk-users list should continue to be used for discussions regarding deployment and usage of Asterisk itself. As we expect the amount of questions and discussions regarding application development and API usage to grow, it makes sense to provide a dedicated forum for those discussions outside of -users and -dev, and this new mailing list serves that purpose. Of course, there's bound to be some cross-over between mailing lists, and sometimes it might not be clear where to post your question. As always, please do not cross post between lists. If your question on asterisk-app-dev is more appropriate for asterisk-users or asterisk-dev, someone will kindly ask you to direct your question to the other mailing lists, where a wider audience may be available to assist you. Thanks - and we all look forward to lots of productive discussions on the new mailing list about building applications that use Asterisk as their communications engine! Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Capture Media IP in CDR (CDR)
On Sun, Oct 13, 2013 at 2:06 PM, CDR wrote: > I need Digium to store this IP in the CDR. I will be honest with the > government and let them know that my tool is incapable of saving lives > or safeguarding our national security because nobody thought about > this. > PD: I am not paying for a patch, since this is huge burden on a small > company like mine, with a single employee, and also because the whole > world will enjoy the benefit. It is not fair that I would have to hire > somebody to patch Asterisk. > I appeal to Digium to patch Asterisk. > I won't comment any further on the technical aspects of what you are looking for; others have already pointed out how various portions of SIP messages can be stored in CDRs and how these portions of the SIP messages are (a) actually of more use than the media IP address in the SDP and (b) meet the requirements being levied by your use case. That aside, I do think it is important to note here that Asterisk does not, by default, have a warranty. This is clearly enumerated in sections 10 and 11 of the GPLv2 license included with Asterisk [1]: NO WARRANTY 11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU. SHOULD THE PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING, REPAIR OR CORRECTION. 12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. END OF TERMS AND CONDITIONS As you are using software licensed free of charge under the GPLv2, there is no obligation by anyone in the community or at Digium to provide you with a patch. If you require assistance, there are many avenues you can choose to pursue to gain such assistance. Just as you profit by running Asterisk, others profit by customizing and supporting the Asterisk project. Asterisk is lucky to have many such talented developers who can assist you with such a development effort. If you really require this functionality, I highly suggest that you look to hire said developers to help you with this feature request [2]. [1] http://svn.asterisk.org/svn/asterisk/branches/11/COPYING [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Capture Media IP in CDR
On Sat, Oct 12, 2013 at 2:19 PM, CDR wrote: > The CHANNEL() function has no idea about the media IP, and also > SIP_HEADER(), since the media IP is not known until the call has been > established and a reinvite has been received and dispatched. I am > using of course, directmedia=yes and directrtpsetup=yes. Hence my > question to the group. > > That information isn't available via a dialplan function or application; about the only place it is communicated is in some log messages when SIP debugging is enabled. I'm rather surprised at this requirement - when you say that your next e-mail will be from "Federal Prison", I'm going to go out on a limb and assume that you're referring to the United States government. Given that Asterisk is widely deployed in the U.S. and recording of remote and/or local media addresses have never been a requirement for CDRs - ever - I have to wonder, what is prompting this request? Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] PJSIP question
On Mon, Sep 23, 2013 at 5:20 AM, CDR wrote: > I am stuck in channel PJSIP trying to see the real flow of SIP > messages, what in regular sip > we used to type "sip set debug on" > Also, is there an automated way to convert sip.conf options to pjsip.conf? > First, please keep in mind that Asterisk 12 is currently in an alpha test period. That means you're operating on what is an early test release of what will become Asterisk 12, so expecting feature parity or perfection is likely to leave you disappointed. The purpose of an alpha is to get feedback on needed features and to help flush out issues before the first full release is made - so expect some bumps along the way. Because the new SIP stack is highly modular, the ability to trace SIP messages is provided by the module res_pjsip_logger. If that module is loaded, all messages sent/received by the stack are displayed on the CLI. Think of 'module load res_pjsip_logger.so' as analogous to 'sip set debug on', and 'module unload res_pjsip_logger.so' as 'sip set debug off'. There is a script in the contrib folder in both the alpha tarball and in the 12 branch that aids in the conversion of a sip.conf configuration file to the pjsip.conf schema. What is currently in the tarball/branch isn't quite complete yet however - Mark has the initial set of conversion options currently up for review on Review Board. If you wanted to make use of that script, I'd highly recommend using the version that is currently up for review, as it will give you more bang for the buck. You can download the patch from here: https://reviewboard.asterisk.org/r/2846/diff/raw/ Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] How to customize CDR(src) value ?
On Thu, Sep 19, 2013 at 9:02 AM, Olivier wrote: > Hi, > > Asterisk 11 doc says CDR(src) value is read-only (see > https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR). > > For various reasons, I would appreciate to change its value so that it my > own presentation rules instead of telco rules. > Very often, I'm connected to telcos through DAHDI (and ISDN). > For instance, telco presents calls with 123456789 while I would prefer a > normalized +34123456789. > > Whenever I change CallerID presentation, the updated value persists in > CDR(callerid) which matches my needs. > Unfortunately, for CDR(dst), I'm still looking for an appropriate function > or application. > > Looking at Asterisk doc, I saw NoCDR and ForkCDR apps but couldn't link > those to what I'm after. > > How can I (re-)set CDR(src) value ? > > You can't. It is a read-only property. If you want a custom value - "my-src" or something like that - you can add a new value to your CDR record by using the CDR function, i.e., Set(CDR(my-src)=+34123456789). Certain CDR backends - such as cdr_custom or cdr_adpative_odbc - have the ability to store custom values. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] DPMA for Asterisk 12?
On Fri, Sep 6, 2013 at 12:43 PM, George Joseph wrote: > Looks like res_digium_phone will need some work for Asterisk 12... > > WARNING[9372]: loader.c:561 load_dynamic_module: Error loading module > 'res_digium_phone.so': > /usr/lib64/asterisk/modules/res_digium_phone.so: undefined symbol: > __ao2_container_alloc > > Quite a lot, actually. Beyond just linking issues, there's that whole new SIP stack thing we'd like to get it using. We're working on it - stay tuned... Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk 12.0.0-alpha1 Now Available!
On Fri, Aug 30, 2013 at 7:18 PM, Matt Behrens wrote: > On Aug 30, 2012, at 5:15 PM, Asterisk Development Team < > asteriskt...@digium.com> wrote: > > > * The Asterisk RESTful Interface (ARI) has been added. This interface > lets > > external systems harness the telephony primitives within Asterisk to > develop > > their own communications applications. Communication with Asterisk is > done > > through a REST interface, while asynchronous events from Asterisk are > > encoded in JSON and sent via a WebSocket. More information on ARI can > be found > > at https://wiki.asterisk.org/wiki/x/lYBbAQ > > > > * Major standardization of the Asterisk Manager Interface and its events > have > > occurred within this version. In particular, the names of Asterisk > channels > > no longer change and are stable throughout the lifetime of the channel. > > More information on the changes in AMI can be seen in the AMI 1.4 > > Specification at https://wiki.asterisk.org/wiki/x/dAFRAQ > > I've been looking occasionally at how 12 work was going and I'm curious > about how AMI and ARI relate. Do they effectively expose the same > functionality, just offering a different style of communication? Or is > there be a reason to prefer one over the other beyond the protocols used? > The general use cases for each protocol are a bit different, although I think some people will find areas of overlap as well. The goal of ARI is to allow for externally controlled communications applications. Unlike AGI or AMI, you wouldn't use ARI to execute dialplan logic or Asterisk applications; you would use ARI to replace a default Asterisk dialplan application with one that performs your own business logic and rules. That's why ARI exposes a lot more of Asterisk's communication primitives and lets you control them in a fine grained fashion - if you want to write your own complex IVR or Queue, you need access to asynchronous media operations, various types of bridges, and control over multiple channels at the same time. AMI, at its heart, is a call control protocol. While you can do some of what AMI does using only ARI, ARI requires handing all of the channels over to the external application through the Stasis dialplan application. If all you used was ARI, you'd lose some of the power of the dialplan. Likewise, while you can do some of what ARI does via a combination of AMI/some AGI variant, the result can be somewhat klunky and difficult to manage - particularly for complex bridging scenarios. Hope that helps! Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk crash issue
sent to a > different port than replies for an existing peer/user. If at all possible, > > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! use the global 'nat' > setting and do not set 'nat' per peer/user. > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! (config > category='S2M-gateway' global force_rport='Yes' peer/user force_rport='No') > > > [Sep 2 16:00:00] WARNING[4970] pbx.c: Extension '_0.', priority 42 in > 'outgoing', label 'UP' already in use at priority 33 > > [Sep 2 16:00:00] WARNING[4970] pbx.c: Extension '_0.', priority 45 in > 'outgoing', label 'DOWN' already in use at priority 36 > > [Sep 2 16:00:01] WARNING[5180] acl.c: Cannot connect > > ** ** > > > It looks like you performed a 'sip reload' during an active call and Asterisk crashed. There have been numerous bug fixes to the 1.8 and later branches to address this kind of issue - since you're running a version of Asterisk 1.8 that is 20 months old, there is a good likelihood that any issue you are facing has already been fixed. Upgrading to a more recent version of 1.8 may be your best course of action. In any case, a log file only showing WARNING messages is often not sufficient for debug a problem, much less a crash. There are explicit instructions on the Asterisk wiki [1] on how to obtain the correct information when Asterisk crashes. If this happens again, please obtain a backtrace using the instructions and file an issue on the Asterisk issue tracker [2]. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira Thanks Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] How to reply with 480 Call-limit to incoming SIP call ?
On Thu, Aug 29, 2013 at 9:07 AM, Gareth Blades <
mailinglist+aster...@dns99.co.uk> wrote:
> On 29/08/13 14:42, Olivier wrote:
>
>> Thanks for your very helpful reply.
>>
>> 1.My system prints out:
>> CLI> core show application Hangup
>>
>> -= Info about application 'Hangup' =-
>>
>> [Synopsis]
>> Hang up the calling channel.
>>
>> [Description]
>> This application will hang up the calling channel.
>>
>> [Syntax]
>> Hangup([causecode])
>>
>> [Arguments]
>> causecode
>> If a is given the channel's hangup cause will be set
>> to the given value.
>>
>> [See Also]
>> Answer(), Busy(), Congestion()
>>
>> How could we improve this Arguments section so that other Asterisk admins
>> can find available values ?
>>
>>
> Have a look in the source code in channels/chan_sip.c and you will see :-
>
> const char *hangup_cause2sip(int cause)
> {
> switch (cause) {
> case AST_CAUSE_UNALLOCATED: /* 1 */
> case AST_CAUSE_NO_ROUTE_**DESTINATION:/* 3 IAX2:
> Can't find extension in context */
> case AST_CAUSE_NO_ROUTE_TRANSIT_**NET:/* 2 */
> return "404 Not Found";
> case AST_CAUSE_CONGESTION: /* 34 */
> case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
> return "503 Service Unavailable";
> case AST_CAUSE_NO_USER_RESPONSE:/* 18 */
> return "408 Request Timeout";
> case AST_CAUSE_NO_ANSWER: /* 19 */
> case AST_CAUSE_UNREGISTERED:/* 20 */
> return "480 Temporarily unavailable";
> case AST_CAUSE_CALL_REJECTED: /* 21 */
> return "403 Forbidden";
> case AST_CAUSE_NUMBER_CHANGED: /* 22 */
> return "410 Gone";
> case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
> return "480 Temporarily unavailable";
> case AST_CAUSE_INVALID_NUMBER_**FORMAT:
> return "484 Address incomplete";
> case AST_CAUSE_USER_BUSY:
> return "486 Busy here";
> case AST_CAUSE_FAILURE:
> return "500 Server internal failure";
> case AST_CAUSE_FACILITY_REJECTED: /* 29 */
> return "501 Not Implemented";
> case AST_CAUSE_CHAN_NOT_**IMPLEMENTED:
> return "503 Service Unavailable";
> /* Used in chan_iax2 */
> case AST_CAUSE_DESTINATION_OUT_OF_**ORDER:
> return "502 Bad Gateway";
> case AST_CAUSE_BEARERCAPABILITY_**NOTAVAIL: /*
> Can't find codec to connect to host */
> return "488 Not Acceptable Here";
> case AST_CAUSE_INTERWORKING:/* Unspecified
> Interworking issues */
> return "500 Network error";
>
> case AST_CAUSE_NOTDEFINED:
> default:
> ast_debug(1, "AST hangup cause %d (no match found
> in SIP)\n", cause);
> return NULL;
> }
>
> For any given hangup cause you can change the sip response there. For a
> list of the hangup numbers and the internal variable name look in
> include/asterisk/causes.h
>
> So if you change chan_sip.c and add the following just before the
> 'AST_CAUSE_NOTDEFINED' line and recompile and reinstall you should in
> theory be able to do a Hangup(44) to achieve what you want.
>
> case AST_CAUSE_REQUESTED_CHAN_**UNAVAIL:/* 44 */
> return "480 Temporarily Unavailable (Call limit)";
>
> Thats only in theory. I havent tested it myself and I am not an asterisk
> developer.
>
>
Also, a table of all of the hangup cause mappings is on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] Am I being hacked?
On Mon, Aug 19, 2013 at 2:29 PM, Eric Wieling wrote: > Actually, you can try enabling the "security" logging destination in > logger.conf. I believe that may contain the info, but it is new in > Asterisk 11. 1.8 and earlier does not have this. > > Nitpick: it was a new feature in Asterisk 10. (Just in case someone is still running that version...) -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Dial application "b" subroutine arguments not passing?
On Fri, Aug 2, 2013 at 12:57 PM, Mitch Claborn wrote:
> Asterisk 11.1.0
>
> I'm trying to use the "b" subroutine of the Dial application so that I can
> do some stuff with our internal applications that need to have access to
> the called channel information. I can see that the subroutine is being
> executed, but the arguments I pass don't see to make it to the subroutine.
>
> [callmenow]
> exten => s,1,NoOp(callmenow: Queue without answer)
> same =>n,Queue(sales,tc)
>
> [dial-to-customer]
> exten => s,1,NoOp(to-customer)
> same =>n,Wait(1)
> same =>n,Playback(custom/callmenow-announce)
> same =>n,GoSub(sub-outbound_caller_id,start,1)
> same =>n,Dial(${TOLL}/${MMCUSTOMER_NUMBER},,*
> b(dial-to-customer-sub,s,1,${MMCUSTOMER_NUMBER},${MEMBERINTERFACE},${MEMBERNAME})
> *)
>
Use a '^' to delineate arguments pass to subroutines. This is actually true
for the U option as well. See:
https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
And:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
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[asterisk-users] Recommended in Asterisk Wiki E-Mail
Hey all - Somehow a mass e-mail option was enabled in Confluence, the software that runs the Asterisk wiki. As a result, if you have an account, you may have gotten a "Recommended in Asterisk Wiki" e-mail. That option shouldn't have been enabled and has now been disabled. You shouldn't receive any more e-mails from the Asterisk Wiki unless you explicitly choose to watch a page. Sorry for the spam! Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] CEL custom variable in outbound channel
On Thu, Jul 18, 2013 at 10:16 AM, Hristo Trendev wrote: > Hi, > > I am using Asterisk 1.8 and trying to pack some custom data in a CEL > HANGUP event. > > In a master (inbound) channel I can set the CHANENL(userfield) to pass > custom information to a CEL event. In the outbound channel created by > Dial() I can also possibly use a macro/gosub on answer and set the > CHANENL(userfield) from there. > > The problem is how to set it in an outbound channel created by Queue() for > example or even one created by Dial(), but which was cancelled or rejected? > > If I read the docs correctly CEL uses a predefined set of values, which > mostly come from the CHANNEL function, but it cannot just pull the value of > any channel variable/function and put it in the event. Also I don't think > that there is any such thing as function value inheritance (setting > __CHANNEL(userfiled) so it is inherited so to say). > > I didn't test it myself, but there is a configuration option in > manager.conf that allows a list of variables to be defined and they will > then be automatically included in AMI events. > > Is there something similar for CEL events (with sqlite backend if that > matters)? Can I just get any channel variable value in a CEL event? I tried > changing the default cel_sqlite3 template to include a channel variable, > but it is always empty. > > So, this is only so helpful, as this solution only applies to Asterisk 11. There may be another way to accomplish this in Asterisk 1.8, but this is the first one that came to my mind - maybe someone else will have another suggestion. If migrating to Asterisk 11 is an option for you, than maybe this will help. In Asterisk 11, you could use a pre-dial handler [1] to apply the userfield directly to the outbound channel on the initial Dial. Because pre-dial handlers are run immediately after channel creation but before dialing or any other action is taken, they work in situations where the dial operation fails or is cancelled. You would have something that looks something like this: exten => s,1,Dial(SIP/foo,,b(default^callee_handler^1)) ... exten => callee_handler,1,NoOp() same => n,Set(CHANNEL(userfield)=my_custom_data) same => n,Return() Note that pre-dial handlers are not directly available in the Queue application. However, if you use Local channel agents, then you could use pre-dial to put the userfield information on the callee SIP channel when the Local channel performs a Dial to the actual SIP device. [1] https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] AMI timeouts
On Thu, Jul 18, 2013 at 3:40 AM, Alexander Frolkin wrote: > Hi, > > > When you execute an Originate action, are you doing so synchronously or > > asynchronously? > > > > A synchronous Originate performs the full outbound dial operation on the > > thread servicing the AMI request. Since each session in AMI gets its own > > thread that services both actions and events, a synchronous Originate can > > block that session from receiving events until it completes. > > I may be misunderstanding, but would your suggestion be to open a new > AMI connection for each action? (Or at least each action that can > block.) > Not at all. The Originate action has a parameter 'async' that, when present and set to True, will not block the thread servicing the AMI session. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Originate I would suggest setting that field to True in your Originate actions. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] AMI timeouts
On Mon, Jul 15, 2013 at 7:59 AM, jg wrote: > When you have many calls, there are usually (read/write=all) a lot of RTP, > RTCP, and VarSet events. This might slow down things, but whether they > occur or not depends on your configuration. > > This might be another thing to look at. > > When you execute an Originate action, are you doing so synchronously or asynchronously? A synchronous Originate performs the full outbound dial operation on the thread servicing the AMI request. Since each session in AMI gets its own thread that services both actions and events, a synchronous Originate can block that session from receiving events until it completes. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?
On Wed, Jul 10, 2013 at 5:35 PM, Mike Diehl wrote: > Hi all, > > I'm contemplating an upgrade from 10.2.4 to 11.4.x. However, the > 1.8.x to 10.4.x upgrade was painful; some of the modules had been > renamed, if I recall correctly. > > So, is there a list of MAJOR changes and GOTCHA's between 10.x and > 11.x? I'm hoping for something a little less granular than the > release notes from 10.2.x to 11.4.x. I don't mind reading, but that > is almost as long as War and Peace! > > Does such a document exist, or do I need to start reading.. > > Upgrade notes: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 While the upgrade notes cover changes to configuration and module status, it is also a good idea to read through what is new: https://wiki.asterisk.org/wiki/display/AST/New+in+11 I wouldn't say it is "War and Peace", but yes, there is some content in there. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] problem with dtmf detection in asterisk 11
On Fri, Jul 5, 2013 at 3:54 PM, wrote: > Hi. I am having problems with asterisk detection dtmf properly in > asterisk 11. I am up to rev 390229. Now, when coming in off a did we > have with Velocity, the dids work fine, but from extensions often it > misses digits -- I can type *4 and it will miss the 4. Often, if I type > quite slowly things will work properly. All dtmf modes are set to > rfc2833. Strangely enough, I did not notice this with asterisk 8, but I > would hate to go back to solve this problem. > > There really shouldn't be any differences between Asterisk 1.8 and Asterisk 11 with respect to DTMF detection - particularly in the RTP engine. There were some issues about 8 months ago with in-band DTMF, but that has been resolved in both LTS versions for quite some time. If you're experiencing issues with RFC 2833 DTMF detection, we'll need some more information to be able to tell what is going on. RTP debug for the endpoint in question would most likely help. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk crash
On Thu, Jul 4, 2013 at 3:30 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:
> Suddenly my asterisk restarted automatically and came up in seven seconds,
>
> While checking core dump I see some message related to snmp.
>
> No symbol table info available.
> #5 0x7fc7e6249faa in agent_thread (arg=) at
> snmp/agent.c:206
> __PRETTY_FUNCTION__ = "agent_thread"
> #6 0x0056dd0b in dummy_start (data=) at
> utils.c:1028
> __cancel_buf = {__cancel_jmp_buf = {{__cancel_jmp_buf = {89647040,
> 7553562169405615537, 140735377460432, 140496194722240, 4, 7,
> -7540143656030687823,
> 7553561768520461745}, __mask_was_saved = 0}}, __pad = {0x7fc7d1c74e90,
> 0x0, 0x0, 0x0}}
> __cancel_arg = 0x7fc7d1c75700
> not_first_call =
> ret =
> a = {start_routine = 0x7fc7e6249eb0 , data = 0x0, name =
> 0x7fc7d1c74d70 "\300\347W\005"}
> #7 0x7fc830e54851 in start_thread () from /lib64/libpthread.so.0
> No symbol table info available.
> #8 0x7fc8323c611d in clone () from /lib64/libc.so.6
> No symbol table info available.
> (gdb) quit
>
> Will this be related to snmp?
>
>
> Possibly, but not necessarily. Without seeing the whole backtrace it's
hard to say for certain.
The Asterisk wiki has instructions on how to properly get a backtrace from
a core dump created by Asterisk:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
Please do file an issue in the issue tracker -
https://issues.asterisk.org- crashes are always bugs.
Thanks!
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] Calls drop after transfer
On Wed, Jul 3, 2013 at 1:31 PM, Carlos Chavez wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > I have an Asterisk 11.4 SIP only system. We are using a SIP trunk > for outside calls. We are having a problem with calls dropping after > a transfer. > > Outside call awswered by phone 101 > 101 transfers to 100 (attended transfer) > call is dropped after a few seconds > > I cannot really think of anything else to check in sip.conf. > Incoming calls never drop if they are not transferred. > > What does Asterisk say when the transfer occurs? You can also look at a trace of the SIP messages during the transfer using 'sip set debug on ' (set it for both the transferer as well as the transfer destination). That should show why the requests are rejected and/or why a call is hungup. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] FTP server down?
On Thu, Jul 4, 2013 at 2:48 AM, A J Stiles wrote: > Not sure if this is the right place to mention it, but . > > The server downloads.asterisk.org was refusing FTP connections last > night, and > still does not seem to be accepting them this morning. > > FTP may not be "modern" or "trendy", but the ability to navigate around > folders textually is nonetheless extremely handy when using a machine with > no > X server, from its own console. > > Thanks for letting us know - we're obviously a little short handed with today being an American holiday and all, but we'll look into it ASAP. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"
c: ...host '192.168.2.18' and port
> '7490'.
> [Jul 2 15:54:44] DEBUG[2737] chan_sip.c: Trying to put 'BYE sip:100'
> onto UDP socket destined for 192.168.2.18:7490
>
>
>
So, these statements show that it is actually using the config file to try
to say the datetime:
[Jul 2 15:54:44] VERBOSE[2737] pbx.c: -- Executing [@avhan:2]
DateTime("SIP/1001-", "136512,,YBd") in new stack
[Jul 2 15:54:44] DEBUG[2737] app_playback.c: string
depth <0>
[Jul 2 15:54:44] DEBUG[2737] app_playback.c: try
in
The DEBUG statements in app_playback indicate the following:
* It will use the configuration in the [en] context (no language was
specified, so it defaults to en)
* It will use the configuration in the datetime extension
* It will attempt a match in the datetime extension on datetime:YBd
* It will attempt to say 201304050530.00-5- 94 based on whatever extension
pattern matches datetime:YBd
Looking at your say.conf config file, you don't have an extension that
matches "datetime". You have one that matches "date" and "time", but not
the combination of the two.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] Endpoint call forwarding
On Tue, Jul 2, 2013 at 2:51 AM, John T. Bittner wrote: > Anyone having issues with endpoint call forwarding on asterisk 11? > > ** ** > > Was working perfect with 10. Issues are not phone related have tried > cisco, polycom and Xlite, all fail. > > ** ** > > Backtrack to 10 and it works ok again. > > ** ** > > Any help is appreciated. > > ** ** > > Thanks > > ** ** > > ** ** > > John Bittner > > CTO > > > I'm not aware of any issues currently open against call forwarding. What specific problems are you seeing? What are the phones sending to Asterisk, and what is Asterisk responding with? A pastebin of a log showing DEBUG and higher level messages when a call forward attempt occurs would help. Thanks Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] CEL logging and queue APP_START/END, maybe an issue?
On Sun, Jun 30, 2013 at 4:24 PM, Fabio Moretti wrote: > Hi, I'm still doing experiment with CEL and recently I've noticed that if: > > 1- call enter in the pbx > 2- call enter a queue > 3- call get answered > 4- the operators call directly another queue, using the queue number > 5- the call get answered > 6- the call end > > the point 4 does not generate a correct logging. In my case I have the > extension 1006 answering the call with linkedid 1371568201.486360, then > call queue number 21. I get only a BRIDGE_UPDATE event: > > '1812621', 'BRIDGE_UPDATE', '2013-06-18 12:11:31', 'ISS23071100', '23071100', > '23071100', '24872939', '', '21', 'from-internal', > 'Local/1006@from-queue-00034a00;2', '', '', '', '', 'Queue', '21,tr,,', '3', > '', '1371568269.486380', '1371568201.486360', > 'Local/714126@from-queue-00034a04;1', '', '', '' > > > here you can se the extension called (21), the source channel > (Local/1006@from-queue-00034a00), the peer > (Local/714126@from-queue-00034a04) and especially you can see that the > app name is "queue". > After this event I have only an APP_END for queue number 21, but no > CHAN_START/END for the peer channel, no queue events, nothing. > This is if I follow the call with linkedid 1371568201.486360. > If I search the uniqueid of the BRIDGE_UPDATE as linkedid > (1371568269.486380) I get the whole queue 21 call history, as I expected to > find in the original call linkedid. > > I think that because the call is the same the linkedid shouldn't change at > all, but in this case the cel logging is generating a "sub" cel, is this by > design? I'm supposed to check if the current event have generate a sub-cel > to reconstruct the call histoery completely? > And if it is by design, isn't a bug put the queue APP_START event in the > sub-cel linkedid and the APP_END in the original linkedid? > > If someone can have a look I've attached the CSV for the two linkedids.. > > Nope, this is entirely expected. A BRIDGE_UPDATE CEL event occurs when a masquerade happens and the participants in a bridge have been altered - that is, a channel came in and pushed one of the channels in the bridge out. In this particular case, Local/714126@from-queue-00034a04;1 has replaced the channel Local/1006@from-queue-00034a00;2 was bridged with, SIP/1006-7705. You can see that SIP/1006-7705 is disposed of immediately following this event. When a BRIDGE_UPDATE happens, you have to start pulling the records from the new channel in with whoever is still in the bridge. This usually means that someone's linkedid changed (as the participants changed). linkedids absolutely do change on a channel in this scenario - when two channels are bridged (which is what has happened when the BRIDGE_UPDATE occurs - the Local channel pushed the SIP channel out), the linkedids are updated on the participants based on who had the older linkedid. In this case, Local/714126@from-queue-00034a04;1's linkedid was updated. So if you want to know everything that happened with that Local channel, you have to tie together both the current linkedid as well as what was its previous linkedid. When you're dealing with CEL, you're operating on a level much closer to what Asterisk is actually doing with its channels. This means having to deal with Local channel pairs and - more importantly - masquerades. This is a whole lot more powerful than CDRs, but does mean that you have to do some bookkeeping to keep track of the channel states. On a side note, the fact that masquerades are hard and tend to require people to do lots of updates was a driving factor in the development efforts that went on in 12. Masquerades are now an implementation detail, so in the future, you won't have to deal with BRIDGE_UPDATE. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Send event/notification from one channel driver ot another
On Sun, Jun 30, 2013 at 11:03 PM, Meadows Hoa wrote: > > Is there a way to send an event notification from one Channel Driver to > another Channel Driver? If I wanted to have one channel driver do something > custom depending on the call state of the other channel driver involved in > the call, how could you send some kind of event or state notification > directly to this other channel driver? > > Yes, although in general this happens within the Asterisk code itself and is not exposed to the dialplan. Typically, if anything (channel driver, resource module, etc.) wants to notify a channel that something has occurred, it queues a control frame on that channel. Control frames include things like media source changes/media updates, indications that signalling actions should take place, etc. What specific use case are you looking at? Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"
On Mon, Jul 1, 2013 at 6:24 AM, Amit Patkar | ATPL wrote: > Hi > > I am using following say.conf file. Its a default file, which comes with > Asterisk installation. > When I call SAY DATETIME AGI function, it simply returns without playing > date & time. Where as if I use mode=old setting, it works. Is this a bug or > mode=new is not implemented for SAY DATETIME AGI function? > > Processing of say.conf is performed by the app_playback module, not by the Asterisk core itself. Do you have the app_playback module loaded? If you run the CLI command "say load", what does it report? Matt -- _ -- 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-11 loop behaviour
On Tue, Jun 25, 2013 at 12:18 PM, James B. Byrne wrote: > > On Tue, June 25, 2013 09:57, Matthew Jordan wrote: > > On Mon, Jun 24, 2013 at 2:37 PM, James B. Byrne > > wrote: > >> It is not an infinite loop but it does go on for an inordinately > >> long time. > >> Does anyone here recognize what is happening and can provide > >> me with an explanation? > >> > > > > Since it is pbx_spool doing the processing, you probably have > > something creating a callfile in /var/spool/asterisk/outgoing > > on startup (or periodically). > > > > I did a quick Google search and found out that this particular context > > is used by FreePBX 2.9's Time Conditions feature - see > > > http://www.freepbx.org/forum/freepbx-distro/distro-discussion-help/odd-failed-calls-in-logs > > for more information. > > > > Thank you. Could I ask what search term you used for google? > > Sure - "freepbx tc-maint" But I did that after I looked at the FreePBX dialplan that came with AsteriskNOW and saw that the context "tc-maint" wasn't in there. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Asterisk-11 loop behaviour
On Mon, Jun 24, 2013 at 2:37 PM, James B. Byrne wrote:
> Arch = x86_64
> OS = CentOS-6.4 (freepbx)
> Asterisk = 11.4.0
> FreePBX = 2.11.0.2
>
> Snom870 Handsets
>
>
> We are in the process of moving to an Asterisk based PBX. At the
> moment most things work as we wish. However, I have just notices that
> when I force a reload using 'amportal a reload' I see this loop start
> in 'asterisk -rvv':
>
>> Channel Local/s@tc-maint-02a4;1 was answered.
>> Launching NoCDR() on Local/s@tc-maint-02a4;1
> [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 attempt_thread:
> Call completed to Local/s@tc-maint
> [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 attempt_thread:
> Call completed to Local/s@tc-maint
> == Spawn extension (tc-maint, s, 5) exited non-zero on
> 'Local/s@tc-maint-02a4;2'
> -- Attempting call on Local/s@tc-maint for application NoCDR()
> (Retry 1)
> -- Executing [s@tc-maint:1] NoCDR("Local/s@tc-maint-02a6;2",
> "") in new stack
> -- Executing [s@tc-maint:2] Set("Local/s@tc-maint-02a6;2",
> "TCMAINT=RETURN") in new stack
> -- Executing [s@tc-maint:3] Gosub("Local/s@tc-maint-02a6;2",
> "timeconditions,1,1()") in new stack
> -- Executing [1@timeconditions:1]
> GotoIfTime("Local/s@tc-maint-02a6;2",
> "08:00-17:00,mon-fri,*,*?truestate") in new stack
> -- Goto (timeconditions,1,9)
> -- Executing [1@timeconditions:9]
> GotoIf("Local/s@tc-maint-02a6;2", "0?falsegoto") in new stack
> -- Executing [1@timeconditions:10]
> ExecIf("Local/s@tc-maint-02a6;2", "0?Set(DB(TC/1)=)") in new
> stack
> -- Executing [1@timeconditions:11]
> Set("Local/s@tc-maint-02a6;2",
> "DEVICE_STATE(Custom:TC1)=NOT_INUSE") in new stack
> -- Executing [1@timeconditions:12]
> ExecIf("Local/s@tc-maint-02a6;2",
> "0?Set(DEVICE_STATE(Custom:TCSTICKY)=INUSE)") in new stack
> -- Executing [1@timeconditions:13]
> GotoIf("Local/s@tc-maint-02a6;2", "0?ext-group,417,1") in new
> stack
> -- Executing [1@timeconditions:14]
> Set("Local/s@tc-maint-02a6;2", "TCSTATE=true") in new stack
> -- Executing [1@timeconditions:15]
> Return("Local/s@tc-maint-02a6;2", "") in new stack
> -- Executing [s@tc-maint:4] System("Local/s@tc-maint-02a6;2",
> "/var/lib/asterisk/bin/schedtc.php 60 /var/spool/asterisk/outgoing
> 1") in new stack
> -- Executing [s@tc-maint:5] Answer("Local/s@tc-maint-02a5;2",
> "") in new stack
>> Channel Local/s@tc-maint-02a5;1 was answered.
>> Launching NoCDR() on Local/s@tc-maint-02a5;1
> [2013-06-24 15:22:32] NOTICE[32681]: pbx_spool.c:402 attempt_thread:
> Call completed to Local/s@tc-maint
> [2013-06-24 15:22:32] NOTICE[32681]: pbx_spool.c:402 attempt_thread:
> Call completed to Local/s@tc-maint
>
> It is not an infinite loop but it does go on for an inordinately long
> time. Does anyone here recognize what is happening and can provide me
> with an explanation?
>
Since it is pbx_spool doing the processing, you probably have something
creating a callfile in /var/spool/asterisk/outgoing on startup (or
periodically).
I did a quick Google search and found out that this particular context is
used by FreePBX 2.9's Time Conditions feature - see
http://www.freepbx.org/forum/freepbx-distro/distro-discussion-help/odd-failed-calls-in-logs
for
more information.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] Questions about sRTP
On Thu, Jun 20, 2013 at 5:10 PM, Mike Diehl wrote: > > > On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp wrote: > >> Mike Diehl wrote: >> >>> Hi all, >>> >>> I'm getting ready to setup SIP/TLS and SRTP. But I have a few >>> questions. The first one is that I was reading an article at: >>> >>> https://supportforums.cisco.com/docs/DOC-15381 >>> >>> That indicated that Asterisk doesn't support TLS as an OPTIONAL >>> transport. It's either all or nothing. Specifically, this is what it >>> said: >>> >> >> Your statement is incorrect. Asterisk supports TLS as an optional >> signaling transport (although if you do SDES SRTP without it then someone >> can snoop on your keys and ultimately decrypt your media). >> >> What it does not support is optional *SRTP*. If a device requests SRTP >> and it's not possible, the call will fail. >> >> > So then, is it safe to say that Asterisk will ALLOW a secure phone call, > but the client hast to REQUEST it? > > I understand that requesting SRTP without SIP/TLS is evil; I just > misunderstood what I was reading. > > I'm also thinking that the AGI script I use to route calls can check if > either leg of a call comes from or goes to port 5061 and play a sound file > to indicate that the cal is 'secure.' Does this seem reasonable? > > You can query a channel using the CHANNEL function ( https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL) to see if the channel currently supports secure communication, and you can request that the outbound channel be made secure using the same function. An example of doing this is on the wiki: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Problem with CEL logging and channel bridging
second half gets the ';2'. When both halves are answered, you know that audio will be forwarded from one half to the other and vice versa. Since you know that DAHDI/i1/96034296-30a3 is in a bridge with Local/1004@from-queue-00019c34;1 and Local/1004@from-queue-00019c34;2* *is in a bridge with IAX2/issuegroup-17175, you automatically know that DAHDI/i1/96034296-30a3 and IAX2/issuegroup-17175 can communicate (at least once everyone has Answered). The system you build on top of CEL has to understand the semantics of Local channels and tie the two together. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] MOH don't work after update
On Sun, Jun 16, 2013 at 2:43 AM, Olivier CALVANO wrote:
>
>
> Hi
>
> we have a small problems.
>
> We have a Asterisk 1.6.1 old server with music on old.
>
> we have updated to AsteriskNow 11.4.0
>
> and now, when we want play sound, we have a errors:
>
> -- Executing [334xx@Accueil_HNO:2]
> BackGround("SIP/SIP05-000c", "Fermeture") in new stack
> [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701
> ast_openstream_full: File Fermeture does not exist in any format
> [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile:
> Unable to open Fermeture (format (alaw)): No such file or directory
> [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180
> pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for
> Fermeture
> -- Executing [334xx@Accueil_Phibee_HNO:4]
> Hangup("SIP/SIP05-000c", "") in new stack
> == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on
> 'SIP/SIP05-000c'
>
>
> I understand that he search the file in .ulaw, but why i don't use the mp3
> ?
>
>
> musiconhold.conf
>
> [default]
> mode=quietmp3
> directory=/var/lib/asterisk/moh
>
> [Horaires]
> mode=quietmp3
> directory=/var/lib/asterisk/moh/Horaires
>
>
>
> ps fax:
> 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G
> asterisk
> 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G
> asterisk -vvvg -c
> 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f
> 8192 Fermeture.mp3
> 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048
> -f 8192 Fermeture.mp3
>
>
> find /var/lib/asterisk/moh/
>
> /var/lib/asterisk/moh/Horaires/Fermeture.mp3
>
> ll
> -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010
> /var/lib/asterisk/moh/Horaires/Fermeture.mp3
>
>
>
>
>
Do you have the format_mp3 module loaded?
Add-on modules are in the addons subdirectory. Typically, these modules are
not built and installed by default, and have to be enabled in menuselect.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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_
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Re: [asterisk-users] Codec Negotiation problem
On Thu, Jun 13, 2013 at 12:04 PM, wrote: > Hi there > > I have asterisk 10.11.1 which seems to have problem negotiating codec. > > Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p > and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw, > h263p. I have tried similar combination of codecs and SIP phone but when > making a video call, it report "Peer doesn't provide video". It seems > Asterisk is failing to set capability correct. Both codecs are enabled on > the SIP Phones > > The 200 OK response from the called XLite phone is declining the video stream: <--- SIP read from UDP:10.10.10.129:48464 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK368135b0;rport=5060 Contact: To: "SAM";tag=0c90cc0c From: ;tag=as24914503 Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, eventlist User-Agent: X-Lite release 4.5.2 stamp 70142 Content-Length: 234 v=0 o=- 13015615910543193 2 IN IP4 10.10.10.129 s=X-Lite 4 release 4.5.2 stamp 70142 c=IN IP4 10.10.10.129 t=0 0 m=audio 53188 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 0 RTP/AVP 115 <-> --- (12 headers 10 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|h263p), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Note that the port for the video stream is set to 0. Asterisk is doing the correct thing: it notes that the answer to its offer declined the video stream, so it disables video for the call between the two endpoints. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Confbridge doesn't kick chan_local
On 06/03/2013 01:03 PM, Chris Gentle wrote: > On Mon, Jun 3, 2013 at 11:52 AM, Matthew Jordan wrote: >> If both (1) and (2) are successful, than there's some impact that the >> Ices application is having on the Local channel that is messing up the >> reference counting inside the ConfBridge. Otherwise, it's an error in >> ConfBridge. > > So what do you think? Should I file a bug? > Please do - https://issues.asterisk.org/jira Please link this conversation on the issue as well, so the bug marshals can reference it when the issue is triaged. http://lists.digium.com/pipermail/asterisk-users/2013-June/279231.html Thanks! Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Is uniqueid/sequence a safe CDR table primary key ?
On 06/03/2013 11:20 AM, Olivier wrote: > Hi, > > When dealing with CDR SQL tables, I always added an auto-incremented > cdr_id key as a primary key, just in case provided uniqueid key went wrong. > > Now I'm facing a situation where I need to insert into a database's > table and from the dialplan, a reference to the CDR record which is > currently processed. > > So my questions are: > > 1. Can uniqueid/sequence (or uniqueid/sequence/calldate) bundle be > safely used as CDR's table primary key (ie I cannot have any > uniqueid/sequence combination from one CDR record to match a past > uniqueid/sequence combination) ? Possibly. Things to keep in mind: * You can run into uniqueid collisions across multiple systems if you do not specify a system name in asterisk.conf or do not specify a unique system name in asterisk.conf. * You can run into uniqueid collisions if your system clock goes backwards for any reason (the uniqueid for a channel happens to use a timestamp for its uniqueness) Whether or not this is unique enough will be completely dependent on your overall system configuration. In general, the recommended combination that *should* uniquely specify a CDR (when configured correctly) is linkedid (which should be enabled and added to your schema), uniqueid, and sequence number, with the asterisk system name specified. -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Confbridge doesn't kick chan_local
On 06/03/2013 08:11 AM, Chris Gentle wrote: > I have a confbridge setup that feeds the conference from the ALSA > microphone input (this is the conference leader) and then uses > app_ices to send the conference audio to icecast. > > I start the conference leader like this: > > console dial 1000_admin@conferences > > I join the ices user to the confbridge with a call file: > > Channel: Local/1000@conferences > MaxRetries: 2 > RetryTime: 60 > WaitTime: 30 > Context: conferences > Extension: 1000_ices > Priority: 1 > > This all seems to work great until I need to kill the conference. My > confbridge user profile sets all users up with end_marked=yes so that > they should be kicked when the leader exits. The Local channel > doesn't seem to exit properly: > That is odd. ConfBridge doesn't really care about the channel technology; if a Local channel joins with the same user profile as a SIP channel, it should treat those channels in the same fashion. Ices may be throwing it off here however. I'd try two different tests: (1) Verify that with all 'normal' channel drivers, such as chan_sip, that the Conference tears down correctly. (2) Verify that a Local channel in an infinite wait gets ejected properly and the conference tears down. You can do that by: exten => 1000_wait,1,Answer() exten => 1000_wait,2,Wait(1) exten => 1000_wait,3,Hangup() (Or something equally sinister) If both (1) and (2) are successful, than there's some impact that the Ices application is having on the Local channel that is messing up the reference counting inside the ConfBridge. Otherwise, it's an error in ConfBridge. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] MeetMe exit status?
On 06/02/2013 08:36 PM, Patrick Lists wrote: > Hi, > > Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I > know for example if a conf ended normally or if someone gave a wrong > conf number or pin? > > Thanks, > Patrick > There is no channel variable that provides that level of granularity. The closest available is the MEETMESECS channel variable, which tells you how many seconds the participant was in the conference. You can find a full list on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/MeetMe+Channel+Variables Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Most suitable version for Production ENV
On 06/01/2013 03:28 PM, Gopalakrishnan N wrote: > Asterisk 1.8 is stable > > On 1 Jun 2013 16:40, "luke devon" <mailto:luke_de...@yahoo.com>> wrote: > > Hi > > As I seen on the Asterisk web site , there is packages called , > > AsteriskLatest Version - 11.4.0 > > asterisk-11-current.tar.gz > > <http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz> > and > > asterisk-1.8-current.tar.gz > > <http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8-current.tar.gz> > > May I now which one is the most suitable for a production environment ? > The Asterisk wiki describes the various versions of Asterisk: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions -- Matthew Jordan Digium, Inc. | Engineering Manager 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] Executing a dynamic sequence of applications
On 05/30/2013 04:46 AM, Grant Bagdasarian wrote: > Hello, > > > > I’m researching the possibilities of multiple communication platforms > like Asterisk and FreeSwitch for handling a dynamic sequence of > applications to execute, like Playback, Read, etc. > > > > This only applies to originating a call from an external application by > using the AMI Manager and the Originate action. > > > > I need to know the following: > > 1) Does the Originate action support multiple Application keys? If > so, how does it handle the order in which they’re added to the Originate > action? No it does not. Originating to an application is a convenience mechanism for when you want to spawn a channel, have a single application execute against it, and hang it up when the application is finished. > 2) If it does not support multiple Application keys, I’ll have to > instruct the Originate action to enter a context in the dialplan, and > pass the sequence of applications in its Variable key. How would I > configure the dialplan context to dynamically handle the sequence of > applications to execute? I was thinking of creating a separate priority > label for each required Application and have each application in the > Variable key routed to the correct priority label. Is this possible? Are > there alternatives for doing what I require? > That's not how the dialplan works. The dialplan is a configured sequence of actions that you take on a channel; it is not meant for external manipulation of the channel. As noted by Yves, external manipulation of channels can be done using AGI. In AGI, the actions executed on a single channel are all performed externally. Your dialplan ends up being quite simple in that case: [default] exten => originated_channel,1,AGI( ... ) And you simply originate your channel to that context, extension, priority tuple. In your particular case, since you're using AMI for call control, you may want to look into AsyncAGI as well. When the channel enters into an AGI application using AsyncAGI, the same AMI connection can be used to control the channel. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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] asterisk-gui-2.1.0-rc1
On 05/25/2013 11:42 AM, luke devon wrote: > Hi Alec, > > I followed that both instructions . It was not success. got the same > issue. I'm not sure the reason is the OS. I am Using CentOS 6.4. > > However, following method was helped me to resolve the problem. > > http://www.digip.org/jansson/releases/jansson-2.4.tar.gz > tar -zxf jansson-2.4.tar.gz > cd jansson-2.4/ > ./configure --prefix=/usr/ && make clean && make && make install > If you needed libjansson, you're running trunk. Trunk is not a stable release branch. This is all fine and good if you're okay with operating on the bleeding edge, but you should know that you aren't running a stable version of Asterisk and to expect some volatility. You should keep up to date with the projects being done in trunk [1], as they'll help you know the areas that are under the most change. You'll also want to watch the asterisk-dev mailing list [2]. If you don't feel comfortable with this, you may be better off basing your work on an LTS release [3], such as Asterisk 1.8 or 11. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Projects [3] http://lists.digium.com/mailman/listinfo/asterisk-dev [4] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 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
