Re: [asterisk-users] Flashing Red Lights in TE420 (5th Gen) Card
Thanks Russ for your response. Finally found time to do more test on this thread. I uninstalled DAHDI-complete 2.9.1.1 and installed an older DAHDI version 2.4.1 It worked! Both READMEs said Digium TE420: PCI-Express quad-port T1/E1/J1 should work. But it seems that 5th gen TE420 (see below) only works with older DAHDI version. 04:08.0 Communication controller: Digium, Inc. Wildcard TE420 quad-span T1/E1/J1 card 3.3V (PCI-Express) (5th gen) (rev 02) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russ Meyerriecks Sent: Saturday, 31 May 2014 3:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Flashing Red Lights in TE420 (5th Gen) Card On Fri, May 30, 2014 at 4:07 AM, Lee, John (Sydney) john@compuware.com wrote: Even without plugging in the ISDN into span 1, all 4 spans are flashing red. Blinking red led is normal for spans which have been configured, but are receiving no signal. I might try plugging up a physical loopback plug to the port to rule out a bad incoming signal. wct4xxp :04:08.0: TE4XXP: Span 1 configured for CCS/HDB3/CRC4 wct4xxp :04:08.0: Span 2 configured for ESF/B8ZS wct4xxp :04:08.0: All spans in alarm : No validspan to source RCLK from This looks like a normal startup for mixed-mode configuration with nothing connected to the ports. I might try setting all spans to T1 or all spans to E1 and plugging one or the other back up to test the connections. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WebRTC / Rejecting secure audio stream errors
I've seen the following appear in some tests with Asterisk 11.11: WARNING[3938][C-0003]: chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101 Specifically, it always happens from a Firefox 24 host but it works without this error from another host running Firefox 26 I did a diff on the SDP and couldn't see anything obviously different except one thing: Firefox 24 only has host candidates for ICE (TURN support was only added in Firefox 25). Is there any way that could cause this error though? It appears the encryption details are sufficient and do not otherwise differ between Firefox 24 and 26: --- ff-24.txt 2014-08-25 15:02:20.452383599 +0200 +++ ff-26.txt 2014-08-25 15:01:42.472346613 +0200 @@ -1,12 +1,12 @@ v=0 -o=Mozilla-SIPUA-24.7.0 14737 0 IN IP4 0.0.0.0 +o=Mozilla-SIPUA-26.0 18111 0 IN IP4 0.0.0.0 s=SIP Call t=0 0 -a=ice-ufrag:301212e4 -a=ice-pwd:d7430f468514f1f2d326d3c944691fbf -a=fingerprint:sha-256 E2:53:6A:FA:6D:E2:3F:7E:24:82:0F:E3:27:34:D1:CC:50:31:42:82:5F:DF:34:9A:4F:42:D1:6D:B7:DB:5C:43 -m=audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101 -c=IN IP4 10.10.1.144 +a=ice-ufrag:2ff98ac6 +a=ice-pwd:dc22648d73c4b421274f31c1953828d4 +a=fingerprint:sha-256 F7:52:A3:46:A4:C3:99:36:83:05:7A:8F:B6:CC:A9:17:0A:45:04:79:3D:D7:F5:39:BE:1D:F3:FF:DA:81:DB:7C +m=audio 51390 UDP/TLS/RTP/SAVPF 109 0 8 101 +c=IN IP4 195.8.117.59 a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 @@ -14,17 +14,21 @@ a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv -a=candidate:0 1 UDP 2113667327 10.10.1.144 54908 typ host -a=candidate:1 1 UDP 2113667327 192.168.1.161 52081 typ host -a=candidate:2 1 UDP 2113667327 195.8.117.161 54978 typ host -a=candidate:0 2 UDP 2113667326 10.10.1.144 58499 typ host -a=candidate:1 2 UDP 2113667326 192.168.1.161 33161 typ host -a=candidate:2 2 UDP 2113667326 195.8.117.161 36491 typ host +a=setup:actpass +a=candidate:0 1 UDP 2122252543 10.10.1.90 60221 typ host +a=candidate:1 1 UDP 1686110207 195.8.117.200 60221 typ srflx raddr 10.10.1.90 rport 60221 +a=candidate:2 1 UDP 8388607 195.8.117.59 51390 typ relay raddr 195.8.117.59 rport 51390 +a=candidate:3 1 UDP 2122187007 192.168.150.1 38505 typ host +a=candidate:0 2 UDP 2122252542 10.10.1.90 55368 typ host +a=candidate:1 2 UDP 1686110206 195.8.117.200 55368 typ srflx raddr 10.10.1.90 rport 55368 +a=candidate:2 2 UDP 8388606 195.8.117.59 51391 typ relay raddr 195.8.117.59 rport 51391 +a=candidate:3 2 UDP 2122187006 192.168.150.1 46478 typ host +a=rtcp-mux - (22 headers 22 lines) --- +--- (22 headers 26 lines) --- Sending to 195.8.117.60:5060 (no NAT) Sending to 195.8.117.60:5060 (no NAT) -Using INVITE request as basis request - kbr110264479udsqistu +Using INVITE request as basis request - hqs8q0vi6pgckcu59a8r Found peer 'example.org' for 'anonymous' from 195.8.117.60:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 109 @@ -35,5 +39,53 @@ Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 -[Aug 25 14:59:29] WARNING[3938][C-0003]: chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101 +Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) +Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) +Peer audio RTP is at port 195.8.117.59:51390 -- _ -- 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] help
Dear all, I was going through sip.conf file and i am not able to understand the working and how to test the functionality of below fields. 1.tcpauthlimit 2.tcpauthtimeout any inputs regarding this will appreciated, thanks in advance Thanks SHIVAKUMAR -- _ -- 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] Understanding local channels
Can someone point me to a good tutorial / explanation of local channels? I've been using them without really understanding what is going on, and we all know how dangerous that is! I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels but I'm just not quite getting it. -- Mitch -- _ -- 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] Understanding local channels
On 25-08-14 17:06, Mitch Claborn wrote: Can someone point me to a good tutorial / explanation of local channels? I've been using them without really understanding what is going on, and we all know how dangerous that is! I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels but I'm just not quite getting it. How about the info on the Asterisk wiki: https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels On the left side there's a menu with examples and modifiers. HTH, Patrick -- _ -- 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] Understanding local channels
On Mon, 25 Aug 2014, Patrick Laimbock wrote: https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels s/displa/display/ -- 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] Understanding local channels
On 8/25/2014 1:33 PM, Patrick Laimbock wrote: On 25-08-14 17:06, Mitch Claborn wrote: Can someone point me to a good tutorial / explanation of local channels? I've been using them without really understanding what is going on, and we all know how dangerous that is! I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels but I'm just not quite getting it. How about the info on the Asterisk wiki: https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels That wiki page isn't REALLY detailed. To what level are you wanting to know more about, Mitch? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FYI: Block Comments
On 8/25/2014 2:36 AM, Brian LaVallee wrote: Hello, Here's a fun issue that recently caused me some serious heartache. Hope this helps others from making the same mistake. Did you know that the configuration parser supports block-comments. Like an idiot, I've been highlighting text between dashes. The configuration parser can do a lot of things. Out of curiosity amongst those reading this - how many of you know about templates? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FYI: Block Comments
The configuration parser can do a lot of things. Out of curiosity amongst those reading this - how many of you know about templates? I use templates and wish the realtime parser would understand them as well.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FYI: Block Comments
On 8/25/14, 11:44 AM, Joshua Colp wrote: On 8/25/2014 2:36 AM, Brian LaVallee wrote: Hello, Here's a fun issue that recently caused me some serious heartache. Hope this helps others from making the same mistake. Did you know that the configuration parser supports block-comments. Like an idiot, I've been highlighting text between dashes. The configuration parser can do a lot of things. Out of curiosity amongst those reading this - how many of you know about templates? I love the idea of templates but since I use realtime database configuration I cannot use them. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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] FYI: Block Comments
I love the templates and used them extensively in the beginning of my asterisk journey. But abandoned them once I went realtime. Moving them to realtime - WOULD BE AWESOME! Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 (Internal: ext. 10015) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Monday, August 25, 2014 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FYI: Block Comments On 8/25/2014 2:36 AM, Brian LaVallee wrote: Hello, Here's a fun issue that recently caused me some serious heartache. Hope this helps others from making the same mistake. Did you know that the configuration parser supports block-comments. Like an idiot, I've been highlighting text between dashes. The configuration parser can do a lot of things. Out of curiosity amongst those reading this - how many of you know about templates? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BEGIN-ANTISPAM-VOTING-LINKS -- Teach CanIt if this mail (ID 01MH4IA5C) is spam: Spam: http://spamaway.npgco.com/canit/b.php?i=01MH4IA5Cm=259b9ad54e59t=2014082 5c=s Not spam: http://spamaway.npgco.com/canit/b.php?i=01MH4IA5Cm=259b9ad54e59t=2014082 5c=n Forget vote: http://spamaway.npgco.com/canit/b.php?i=01MH4IA5Cm=259b9ad54e59t=2014082 5c=f -- END-ANTISPAM-VOTING-LINKS -- _ -- 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] FYI: Block Comments
On Mon, 25 Aug 2014, Joshua Colp wrote:
how many of you know about templates?
(You may get more replies with a more 'on-target' subject. I lost
interest in 'block comments' but was curious why the thread was still
getting replies.)
Love templates. Use them in extensions.conf, sip.conf, and iax.conf every
day.
Here's an example from extensions.conf:
[party-line](digit-timeout,h,i,max-timeout,pound-main,s)
same = n, agi(write-cdr)
same = n, background(${PROMPTS-PATH}/0116)
...
Where the templates look like:
[digit-timeout](!)
exten = t,1,goto(${CONTEXT},s,1)
[h](!)
exten = h,1,goto(finish-call,h,1)
[i](!)
exten = i,1,goto(${CONTEXT},s,1)
[max-timeout](!)
exten = T,1,goto(max-time,s,1)
[pound-main](!)
exten = #,1,goto(main-menu,s,1)
[s](!)
exten = s,1,verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
Note that the 's' template has to be the last template specified in the
template list. Also, that '${EXTEN}@${CONTEXT}' makes for a quick
cut-n-paste into the 'dialplan show' CLI command.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] FYI: Block Comments
Holy damn - I didn't know you could use templates in extensions! Mind =
Blown.
Matt Hoskins | NPG Corp | Systems Architect
816.749.2815 (Internal: ext. 10015)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: Monday, August 25, 2014 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FYI: Block Comments
On Mon, 25 Aug 2014, Joshua Colp wrote:
how many of you know about templates?
(You may get more replies with a more 'on-target' subject. I lost interest
in 'block comments' but was curious why the thread was still getting
replies.)
Love templates. Use them in extensions.conf, sip.conf, and iax.conf every
day.
Here's an example from extensions.conf:
[party-line](digit-timeout,h,i,max-timeout,pound-main,s)
same = n, agi(write-cdr)
same = n, background(${PROMPTS-PATH}/0116)
...
Where the templates look like:
[digit-timeout](!)
exten = t,1,goto(${CONTEXT},s,1)
[h](!)
exten = h,1,goto(finish-call,h,1)
[i](!)
exten = i,1,goto(${CONTEXT},s,1)
[max-timeout](!)
exten = T,1,goto(max-time,s,1)
[pound-main](!)
exten = #,1,goto(main-menu,s,1)
[s](!)
exten = s,1,
verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
Note that the 's' template has to be the last template specified in the
template list. Also, that '${EXTEN}@${CONTEXT}' makes for a quick
cut-n-paste into the 'dialplan show' CLI command.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Forget vote:
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Re: [asterisk-users] Understanding local channels
On Mon, Aug 25, 2014 at 11:33 AM, Patrick Laimbock patr...@laimbock.com wrote: On 25-08-14 17:06, Mitch Claborn wrote: Can someone point me to a good tutorial / explanation of local channels? I've been using them without really understanding what is going on, and we all know how dangerous that is! I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels but I'm just not quite getting it. How about the info on the Asterisk wiki: https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels On the left side there's a menu with examples and modifiers. HTH, Patrick It may also help to check out the section on Channels: https://wiki.asterisk.org/wiki/display/AST/Channels Before going into the Local Channel config section:https://wiki.asterisk.org/wiki/display/AST/Local+Channel If you can think of a way we can improve the documentation on Local Channels, let us know. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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
On Fri, Aug 22, 2014 at 6:48 AM, chandapure shiva chandapure.shiv...@gmail.com wrote: Dear all, I was going through sip.conf file and i am not able to understand the working and how to test the functionality of below fields. 1.tcpauthlimit 2.tcpauthtimeout any inputs regarding this will appreciated, thanks in advance Do you have a specific question? What do you mean How to test the functionality of below fields? Here is the documentation on those options from the sip.conf sample file: ;tcpauthtimeout = 30; tcpauthtimeout specifies the maximum number ; of seconds a client has to authenticate. If ; the client does not authenticate beofre this ; timeout expires, the client will be ; disconnected. (default: 30 seconds) ;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of ; unauthenticated sessions that will be allowed ; to connect at any given time. (default: 100) -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Can't hangup channel from CLI
On Fri, Aug 22, 2014 at 6:00 PM, Steve Edwards asterisk@sedwards.com wrote: Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting Asterisk from a Tekelec T9000. I'm accumulating stuck channels. snip I haven't identified what callers are doing to reproduce the error reliably yet. Any clues or suggestions? You might see if they are getting stuck due to a deadlock: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace?src=search#GettingaBacktrace-GettingInformationForADeadlock If you get the traces required, you could open an issue on the bug tracker. If commands like core show channel channel and sip show channel channel work then you'll want to attach that data as well. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Understanding local channels
Here's my current specific scenario. I have a working call me now
solution on our web site. The customer types in their phone number, it
goes into our normal sales asterisk queue via an AMI action. When the
agent answers the call, he gets a brief announcement then asterisk dials
the customer's number. (This works in Asterisk 11. There is an
apparent bug in asterisk 12 with queue variables:
https://issues.asterisk.org/jira/browse/ASTERISK-24267) It works, but
I'm struggling to understand how.
*AMI Action:*
Action: Originate
Channel: Local/s@callmenow/n
Context: dial-to-customer
Exten: s
Priority: 1
Async: true
Variable: MMCALLMENOWID=107
Timeout: 99
Callerid: Call Me Now 778
*Dial Plan:*
[callmenow]
exten = s,1,NoOp(callmenow: Queue without answer)
same =n,Queue(sales,Rtc)
[dial-to-customer]
exten = s,1,NoOp(dial-to-customer channel=${CHANNEL(name)})
same =n,Wait(1)
same =n,Playback(custom/callmenow-announce)
; do some more stuff
same
=n,Dial(${TOLL}/${MMCUSTOMER_NUMBER},,TKU(dial-to-cust-connect-sub))
Mitch
On 08/25/2014 11:43 AM, Joshua Colp wrote:
On 8/25/2014 1:33 PM, Patrick Laimbock wrote:
On 25-08-14 17:06, Mitch Claborn wrote:
Can someone point me to a good tutorial / explanation of local
channels? I've been using them without really understanding what is
going on, and we all know how dangerous that is!
I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels
but
I'm just not quite getting it.
How about the info on the Asterisk wiki:
https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels
That wiki page isn't REALLY detailed. To what level are you wanting to
know more about, Mitch?
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[asterisk-users] cmd Dial with U option
In my dialplan, when I dial 99 it rings SIP/2000
When SIP/2000 answers, it hears 9 every 5 seconds until someone dials 9,
what makes 2 legs been bridged.
My problem is: If I hangup, SIP/2000 continues to hears 9 until someone
dials 9 - it not stops
If SIP/2000 hangup - then the call is ended - what is OK
Is there some workaround? I was thinking in use G option - however I don't
figured out yet how
[TesteU]
exten = s,1,noop()
exten = s,n(READ),read(OPTION,digits/9,1,s,1,5)
exten = s,n,noop(${OPTION})
exten = s,n,GotoIf($[${OPTION} = 9]?END)
exten = s,n,Goto(READ)
exten = s,n(END),noop()
[default]
exten = 99,1,dial(sip/2000,,U(TesteU^s^1))
Thanks
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