Re: [asterisk-users] AGI
Please don't top post. On Fri, 15 Mar 2013, Gustavo Salvador wrote: The reason I have not to use an AGI library and also use bash shells is because the asterisk box I'm working on, have not AGI library, there is installed on a basic first version of Centos, and it is not an asterisk pure box. It's from a vendor, and also have some annoyances specially with the settings and configuration files. And I can not upgrade it, because it is running production. By the way, I have just get what I need. The bash shell if anyone is interested is: #!/usr/bin/bash echo "EXEC \"Dial\" \"DAHDI/g2/$1\"" read line Can we at least pretend to follow the protocol? :) #!/bin/bash # discard the AGI variables while [ -s "$(read)" ] do : done # dial the number from the command line echo "exec dial dahdi/g2/$1" # ignore the response read # (end of agiFile.agi) At least this way, the 'next guy' won't pull out his hair wondering why the response to 'dial' is 'agi_request: agiFile.agi' BTW, you do not have to name an AGI 'example.agi'. 'example.sh' or just 'example' will do just fine. If you don't have PHP or Perl, how about C? You could always compile on a VM on another box. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_rtsp.c ported to Asterisk 11.x
Hey, Can you send me URL to download the tar ball pls? Mitul Limbani On Saturday, March 16, 2013, Robert Krakora wrote: > Hi, > > If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I > have tested it with GStreamer RTSP server and a C920 webcam streaming H264 > SVC video from one machine to another machine running Linphone. Contact me > at this e-mail address robkrak...@messagenetsystems.com 'cvml', 'robkrak...@messagenetsystems.com');>for source code. > > Best Regards, > > -- > Rob Krakora > MessageNet Systems > 101 East Carmel Dr. Suite 105 > Carmel, IN 46032 > (317)566-1677 Ext 212 > (317)663-0808 Fax -- Regards, Mitul Limbani, Chief Architech & Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 -- _ -- 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] 回覆︰ app_rtsp.c ported to Asterisk 11.x
Dear Sir, I want the source code . Please email to me ... Thank Regard/chui king man > > 寄件人︰ Robert Krakora >收件人︰ asterisk-users@lists.digium.com; asterisk-...@lists.digium.com >傳送日期︰ 2013年03月16日 (週六) 4:10 AM >主題︰ [asterisk-users] app_rtsp.c ported to Asterisk 11.x > > >Hi, > >If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have >tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC >video from one machine to another machine running Linphone. Contact me at >this e-mail address robkrak...@messagenetsystems.com for source code. > >Best Regards, > >-- >Rob Krakora >MessageNet Systems >101 East Carmel Dr. Suite 105 >Carmel, IN 46032 >(317)566-1677Ext 212 >(317)663-0808 Fax >-- >_ >-- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_rtsp.c ported to Asterisk 11.x
Hi, If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC video from one machine to another machine running Linphone. Contact me at this e-mail address robkrak...@messagenetsystems.com for source code. Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext 212 (317)663-0808 Fax -- _ -- 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] Call blocking issue
On 03/15/13 13:56, Jewel Nuruddin wrote:
sorry if this is already asked?
May get any help about call blocking issue
Asterisk Version : Asterisk 1.6.2.7
FreePBX Version : 2.7.0.10
OS: CentOS
My asterisk is running basic incomming and outgoin call is working but
I can not configure call blocking and IVR.
any one can give me some hitns?
Thanks
Jewel
See:
http://www.voip-info.org/wiki/view/Asterisk+func+BLACKLIST
The solution worked for me:
exten => s,1,Wait(1)
exten => s,n,GotoIf(${BLACKLIST()}?blacklisted,s,1)
...
[blacklisted]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Playback(privacy-you-are-blacklisted)
exten => s,n,Hangup
--
Joseph
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Re: [asterisk-users] PRI Called Party Number Info
> 14.03.2013 17:53, Gianluca Merlo wrote:
> > Hello Grigoriy,
> >
> > i think that you can access the information you need by using the
> > dialplan function CALLERID(num-plan). It should contain the lower 7
> > bits of the Q.931 type-of-number/numbering-plan-identification
> > octet.
> >
> > Best regards
> >
> > Gianluca
> >
> >
>
> Nope. CALLERID(num-plan) shows numbering plan of caller's number, not
> the called one. In example, where A calls B, CALLERID(num-plan) shows
> numbering plan of A number. In my case I need to know numbering plan
> of
> B number.
>
> CALLERID(dnid-num-plan) is not set anywhere for PRI calls.
>
> Looks like CALLEDTON is the right answer, however it needs mangling
> (${MATH(${CALLEDTON}>>4&0x7,i)}) to get right values.
I created the following issue to get CALLERID(dnid-num-plan) to have the
same value as CALLEDTON.
https://issues.asterisk.org/jira/browse/ASTERISK-21248
Richard
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Re: [asterisk-users] Asterisk does not persist callgroup and pickgroup configuration.
I think you maybe need to ask on elastix forum. They have a lot of files whit "on the fly creation". Yo need to see if this is a bug or another question. When you reload asterisk from the frontend you will lose every change in those files. The only files who the reload don't recreate are those who hace "custom" in the filename. Best regards. Emiliano Emiliano Vazquez | PcCentro S.R.L. Office: +54 (11) 4635-7764 ext. 4 Celular: 15.6253.7165 Mail: emilianovazq...@gmail.com Web: http://www.pccentro.com.ar -Original Message- From: "Luis H. Forchesatto" Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 15 Mar 2013 09:49:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk does not persist callgroup and pickgroup configuration. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI
Hi,
Thank you. The reason I have not to use an AGI library and also use bash shells
is because the asterisk box I'm working on, have not AGI library, there is
installed on a basic first version of Centos, and it is not an asterisk pure
box. It's from a vendor, and also have some annoyances specially with the
settings and configuration files. And I can not upgrade it, because it is
running production.
By the way, I have just get what I need.
The bash shell if anyone is interested is:
#!/usr/bin/bash
echo "EXEC \"Dial\" \"DAHDI/g2/$1\""
read line
AGI invocation:
AGI(agiFile.agi,${CALLERID(num)})
Regards,
Gustavo
El 14/03/2013, a las 17:24, Steve Edwards escribió:
> (A more specific subject may yield better answers -- better bait == better
> fish.)
>
> On Thu, 14 Mar 2013, Gustavo Salvador wrote:
>
>> Does any one knows how to place a call from a shell agi? I guess is
>> something like echo Exec Dial(DAHDI/g2/2010,,W).
>
> While you can write an AGI in any language that has reasonable STDIN/STDOUT
> facilities (fortran was a little bit frustrating), a shell language like bash
> would not be my first choice.
>
> Is there any reason you are not open to using a more 'mainstream' scripting
> language like Perl or PHP? You'll find more people with that experience
> willing to help.
>
> My personal favorite is to use C because I know it best and because it is a
> compiled language so your AGIs execute 'instantaneously.'
>
> Whatever language you choose, use an established library for the AGI
> protocol. Nobody gets it right the first time.
>
>> Algo how i get the dnid variable?
>
> This implies that you need to read up a bit on the AGI protocol. The second*
> thing an AGI should to is read the AGI variables -- one of which is agi_dnid.
>
> You can pass command line parameters to your AGI when you execute the agi()
> function in your dialplan. For example:
>
>exten = *,n,agi(my-first-agi,${AGENT-ID})
>
> Your AGI can then access the command line arguments as you would
> normally expect for a program executed from a shell -- $argv[1],
> $argv[2], etc for PHP.
>
> Personally, I dislike 'positional' parameters. I prefer to use the
> getopt_long() function so my dialplan is 'self-documenting' and easier
> to maintain so for me, the same command would look like:
>
>exten = *,n,agi(my-first-agi,--agent-id=${AGENT-ID})
>
> The difference is not all that apparent with a single parameter, but if you
> have a half-dozen, it is much more obvious and once you've make the change,
> you will always use it for consistency.
>
> *) The first thing an AGI should do is set up a signal handler to catch
> 'SIGHUP' if the channel is hung up prematurely.
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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[asterisk-users] Asterisk uses 3 seconds to send ACK after OK
Hello! We recently upgraded one of our customers from 1.4.44 to 1.8.15-cert1. We have several other customers running both versions. The customer in question does not use us as their provider as they’re located in a different country. When they make outgoing calls, there is a 3 second delay between answering the call and the call being established. When debugging this, I found that Asterisk waits 3 seconds after receiving 200 OK before returning the ACK. See attached image. There’s no verbose output in the CLI during this time. I turned on full debugging. This seems to produce around a hundred lines of debug per second until suddenly I see a full 3 seconds stop just before sending the ACK. [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:;branch=z9hG4bK135effb0 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 2 [ 66]: From: "a" ;tag=as6b9fcf86 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 3 [ 61]: To: ;tag=1014243474 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 4 [ 59]: Call-ID: 03dffd7b5ecd7eb47c2bae6b101ba1aa@62.109.37.34:5088 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 6 [ 51]: Contact: [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 7 [117]: Record-Route: [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 8 [ 69]: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 9 [ 24]: Supported: timer, 100rel [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 11 [ 19]: Content-Length: 352 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 12 [ 0]: [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 0 [ 3]: v=0 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 1 [ 35]: o=- 25469059 0 IN IP4 ccc.ccc.ccc.ccc [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 2 [ 13]: s=Cisco SDP 0 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 3 [ 22]: c=IN IP4 ccc.ccc.ccc.ccc [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 4 [ 5]: t=0 0 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 5 [ 31]: m=audio 21252 RTP/AVP 8 101 100 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 6 [ 33]: a=rtpmap:101 telephone-event/8000 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 7 [ 15]: a=fmtp:101 0-15 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 8 [ 23]: a=rtpmap:100 X-NSE/8000 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 9 [ 18]: a=fmtp:100 200-202 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 10 [ 9]: a=X-sqn:0 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 11 [ 28]: a=X-cap: 1 audio RTP/AVP 100 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 12 [ 33]: a=X-cpar: a=rtpmap:100 X-NSE/8000 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 13 [ 28]: a=X-cpar: a=fmtp:100 200-202 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c:Body 14 [ 26]: a=X-cap: 2 image udptl t38 [Mar 15 13:16:05] VERBOSE[27947] chan_sip.c: [Mar 15 13:16:05] --- (12 headers 15 lines) --- [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: = Looking for Call ID: 03dffd7b5ecd7eb47c2bae6b101ba...@xxx.xxx.xxx.xxx: (Checking To) --From tag as6b9fcf86 --To-tag 1014243474 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Acked pending invite 102 [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Stopping retransmission on '03dffd7b5ecd7eb47c2bae6b101ba...@xxx.xxx.xxx.xxx:' of Request 102: Match Found [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: SIP response 200 to standard invite [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: SIP response 200 to standard invite [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Call 03dffd7b5ecd7eb47c2bae6b101ba...@xxx.xxx.xxx.xxx: responded to our reinvite without changing SDP version; ignoring SDP. [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Updating call counter for outgoing call [Mar 15 13:16:05] DEBUG[27947] chan_sip.c: build_route: Record-Route hop: [Mar 15 13:16:05] VERBOSE[27947] chan_sip.c: [Mar 15 13:16:05] list_route: hop: [Mar 15 13:16:05] DEBUG[27947] netsock2.c: Splitting 'FQDNz:5060' into... [Mar 15 13:16:05] DEBUG[27947] netsock2.c: ...host 'FQDNz' and port '5060'. [Mar 15 13:16:05] DEBUG[27931] devicestate.c: No provider found, checking channel drivers for SIP – FQDNy [Mar 15 13:16:05] DEBUG[27931] chan_sip.c: Checking device state for peer FQDNy [Mar 15 13:16:05] DEBUG[27931] devicestate.c: Changing state for SIP/FQDNy - state 2 (In use) [Mar 15 13:16:05] DEBUG[27931] devicestate.c: device 'SIP/FQDNy' state '2' [Mar 15 13:16:05] DEBUG[27967] app_queue.c: Device 'SIP/FQDNy' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Mar 15 13:16:08]
Re: [asterisk-users] Disagreements between codec_siren14 and Polycom sources
I'm answering my own email here: > There appears to be a disagreement between the encoding given in the > sources for Siren14 that are downloaded from Polycom (and the ITU, both > are the same) and that implemented by codec_siren14.so. The latter > agrees with the actual device. The disagreement is in byte-swapping of the encoded stream. Once that's done, things work fine. If anybody wants a codec that can transcode between Siren14 and slin32 (which is better than Digium's codec_siren14 codec which goes to slin and slin16), let me know. I can send a file that calls the Polycom/ITU code. -- _ -- 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 does not persist callgroup and pickgroup configuration.
Freepbx is over writing the conf files ? On Mar 15, 2013 8:11 AM, "Luis H. Forchesatto" wrote: > Greetings. > > I'm running asterisk here (elastix) and I have a few extensions configured > in it. I have here two different callgroup/pickgroup where the extensions > are configured in, but it doesn't work when I try do pickup a call. Looking > the config file (sip_additional.conf) I see they are not configured with > callgroup/pickgroup, the fields are empty. > > Manually inserting callgroup/pickgroup on the extensions worked just fine > but the next day the configuration just vanished and the extensions was not > working. > > Has someone a clue of whats going on here? > > -- > Att.* > *** > Luis H. Forchesatto > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk does not persist callgroup and pickgroup configuration.
Greetings. I'm running asterisk here (elastix) and I have a few extensions configured in it. I have here two different callgroup/pickgroup where the extensions are configured in, but it doesn't work when I try do pickup a call. Looking the config file (sip_additional.conf) I see they are not configured with callgroup/pickgroup, the fields are empty. Manually inserting callgroup/pickgroup on the extensions worked just fine but the next day the configuration just vanished and the extensions was not working. Has someone a clue of whats going on here? -- Att.* *** Luis H. Forchesatto -- _ -- 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] Disagreements between codec_siren14 and Polycom sources
On 03/15/2013 10:41 AM, Richard Kenner wrote: There appears to be a disagreement between the encoding given in the sources for Siren14 that are downloaded from Polycom (and the ITU, both are the same) and that implemented by codec_siren14.so. The latter agrees with the actual device. If I make a .sln32 file and run the encoder from ITU/Polycom with encode 0 foo.sln32 foo.siren14 48000 14000 the resulting file doesn't play back correctly with the Digium's siren14 codec. I know the parameters are correct because the file is the same size as that made by the Digium codec. Both sets of decoders/encoders (Digium and Polycom/ITU) are symmetric and can decode what they encode, but neither can read the encoding of the other. Is there some subtle difference between G.722.1C and Siren14? G,722.1C is not the same as Siren 14. This is stated in the Polycom material but they don't really indicate how different the two are. More importantly, they are vague about whether the two can be expected to interwork satisfactorily. Polycom only offer source code for G.722.1C, so you can't really figure out the differences for yourself. People are really sloppy about these names and something called siren14 might well be G.722.1C. I assume something called G.722.1C is always G.722.1C. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Called Party Number Info
14.03.2013 17:53, Gianluca Merlo wrote:
Hello Grigoriy,
i think that you can access the information you need by using the
dialplan function CALLERID(num-plan). It should contain the lower 7
bits of the Q.931 type-of-number/numbering-plan-identification octet.
Best regards
Gianluca
Nope. CALLERID(num-plan) shows numbering plan of caller's number, not
the called one. In example, where A calls B, CALLERID(num-plan) shows
numbering plan of A number. In my case I need to know numbering plan of
B number.
CALLERID(dnid-num-plan) is not set anywhere for PRI calls.
Looks like CALLEDTON is the right answer, however it needs mangling
(${MATH(${CALLEDTON}>>4&0x7,i)}) to get right values.
--
Best regards,
Grigoriy
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