Re: [asterisk-users] Make phone ring through webserver using Asterisk
On Saturday 16 November 2013, akhilesh chand wrote: What is the easiest way? And how can it be implemented? i don't care if i can hear something, it's enough that it rings Just inject a callfile into /var/spool/asterisk/outgoing/ . One end is the extension you want to ring, the other end is a dummy extension in a special context which (optionally) can play music down the line. Just do it exactly like an alarm call, but without using `touch` to set a future timestamp on the callfile. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CONNECTEDLINE and panasonic 500
On Mon, Nov 18, 2013 at 1:21 AM, Dmitry Melekhov d...@belkam.com wrote:
Hello!
I have following connections over isdn pri:
avaya definity---pri--asterisk--pri-panasonic 500
Just because panasonic 500 can't send user's names.
I also want to have reverse callerid for avaya users.
But if there is no answer in dial plan:
exten = _,1,Set(CONNECTEDLINE(name)=${DB(names/${EXTEN})})
;exten = _,n,Answer
exten = _,n,Dial(DAHDI/g4/${EXTEN})
exten = _,n,Hangup
there is no name on avaya display after panasonic user pick up...
With answer name appears immediately but this is not what we want :-(
You need to use the 'I' Dial option to inhibit the connected line date from
the
Dial from overwriting the values setup by your dialplan. See [1].
Richard
[1]
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
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[asterisk-users] app_swift on centos 6 X64
Hi is a list could be off topic ;) , but someone has installed the latest version of app_swift on centos 6 for asterisk 1.8 I'm trying to make with this manual, but have had no success http://www.cepstral.com/en/support/telephony/faq?os=linuxsection=getting-started gcc -I/opt/swift/include -I/usr/include -g -Wall -fPIC -D_SWIFT_VER_6 -D_AST_VER_1_8 -c -o app_swift.o app_swift.c app_swift.c:36:19: error: swift.h: No such file or directory app_swift.c:38:38: error: swift_asterisk_interface.h: No such file or directory app_swift.c:163: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before âswift_cbâ app_swift.c: In function âapp_execâ: app_swift.c:322: error: âswift_engineâ undeclared (first use in this function) app_swift.c:322: error: (Each undeclared identifier is reported only once app_swift.c:322: error: for each function it appears in.) app_swift.c:322: error: âengineâ undeclared (first use in this function) app_swift.c:323: error: âswift_portâ undeclared (first use in this function) app_swift.c:323: error: âportâ undeclared (first use in this function) app_swift.c:324: error: âswift_voiceâ undeclared (first use in this function) app_swift.c:324: error: âvoiceâ undeclared (first use in this function) app_swift.c:325: error: âswift_paramsâ undeclared (first use in this function) app_swift.c:325: error: âparamsâ undeclared (first use in this function) app_swift.c:326: error: âswift_result_tâ undeclared (first use in this function) app_swift.c:326: error: expected â;â before âsresultâ app_swift.c:327: error: âswift_background_tâ undeclared (first use in this function) app_swift.c:327: error: expected â;â before âtts_streamâ app_swift.c:363: warning: implicit declaration of function âswift_engine_openâ app_swift.c:368: warning: implicit declaration of function âswift_params_newâ app_swift.c:369: warning: implicit declaration of function âswift_params_set_stringâ app_swift.c:382: warning: implicit declaration of function âswift_port_openâ app_swift.c:393: warning: implicit declaration of function âswift_register_ast_chanâ app_swift.c:403: warning: implicit declaration of function âswift_port_set_voice_by_nameâ app_swift.c:410: error: âSWIFT_EVENT_AUDIOâ undeclared (first use in this function) app_swift.c:410: error: âSWIFT_EVENT_ENDâ undeclared (first use in this function) app_swift.c:410: error: âSWIFT_EVENT_ERRORâ undeclared (first use in this function) app_swift.c:415: warning: implicit declaration of function âswift_port_set_callbackâ app_swift.c:415: error: âswift_cbâ undeclared (first use in this function) app_swift.c:417: warning: implicit declaration of function âSWIFT_FAILEDâ app_swift.c:417: warning: implicit declaration of function âswift_port_speak_textâ app_swift.c:417: error: âtts_streamâ undeclared (first use in this function) app_swift.c:574: error: âsresultâ undeclared (first use in this function) app_swift.c:574: warning: implicit declaration of function âswift_port_stopâ app_swift.c:574: error: âSWIFT_EVENT_NOWâ undeclared (first use in this function) app_swift.c:622: warning: implicit declaration of function âswift_port_closeâ app_swift.c:625: warning: implicit declaration of function âswift_engine_closeâ make: *** [app_swift.o] Error 1 ## later with this and not work http://www.cepstral.com/code/appbugcode.txt any idea? -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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
[asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?
Took me a while but I have finally embraced cloud computing and all the benefits. The only thing I have yet to feel comfortable about putting in the cloud is real live Asterisk boxes to be used in production. I know it's being done because as far as I know Twilio is using Amazon for their Asterisk boxes. I have read all the fun articles on building hobby type systems and that's all great. What I really need to hear is from those that have deployed Asterisk in Amazon or Digital Ocean and how many simultaneous calls they are pushing through it and what the call quality and reliability has been. Right now I am still using dedicated hardware but I could become much more redundant and scale much faster using Amazon or Digital Ocean. Thanks in advance for any information from those that have already been down this road... -- _ -- 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] CONNECTEDLINE and panasonic 500
18.11.2013 20:51, Richard Mudgett пишет:
On Mon, Nov 18, 2013 at 1:21 AM, Dmitry Melekhov d...@belkam.com
mailto:d...@belkam.com wrote:
Hello!
I have following connections over isdn pri:
avaya definity---pri--asterisk--pri-panasonic 500
Just because panasonic 500 can't send user's names.
I also want to have reverse callerid for avaya users.
But if there is no answer in dial plan:
exten = _,1,Set(CONNECTEDLINE(name)=${DB(names/${EXTEN})})
;exten = _,n,Answer
exten = _,n,Dial(DAHDI/g4/${EXTEN})
exten = _,n,Hangup
there is no name on avaya display after panasonic user pick up...
With answer name appears immediately but this is not what we want :-(
You need to use the 'I' Dial option to inhibit the connected line date
from the
Dial from overwriting the values setup by your dialplan. See [1].
Richard
[1]
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
Hello!
Thank you!
It works :-)
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Re: [asterisk-users] CEL for attented transfer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nobody, really? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 Le 17/11/2013 19:03, Jean-Denis Girard a écrit : Hi list, I'm trying to use CEL to display channel information in real time. It works fine for simple calls, blind transfers, or SIP attended transfers (initiated from the SIP phone). My problem is for Asterisk attended transfers (atxfer as configured in features.conf). The scenario is: . phone 107 calls phone 100, . 100 dials the atxfer code, . 107 is on hold, and 100 hears the transfer message, . 100 dials phone 103, . 103 answers, . 100 hangups, . 107 and 103 are connected, . 107 hangups. CEL is configured with apps=all and events=ALL, and events are stored in a database via cel_pgsql. This is the list of events in the database for this call: eventtype |channame| peer - -++--- CHAN_START | SIP/107-0274 | CHAN_START | SIP/100-0275 | ANSWER | SIP/100-0275 | ANSWER | SIP/107-0274 | BRIDGE_START | SIP/107-0274 | SIP/100-0275 CHAN_START | Local/103@100-0042;1 | CHAN_START | Local/103@100-0042;2 | CHAN_START | SIP/103-0276 | ANSWER | SIP/103-0276 | ANSWER | Local/103@100-0042;2 | BRIDGE_START | Local/103@100-0042;2 | SIP/103-0276 ANSWER | Local/103@100-0042;1 | BRIDGE_START | SIP/100-0275 | Local/103@100-0042;1 BRIDGE_END | SIP/100-0275 | Local/103@100-0042;1 ATTENDEDTRANSFER | SIP/107-0274 | Local/103@100-0042;1 CHAN_START | Transfered/SIP/107-0274| BRIDGE_END | Transfered/SIP/107-0274ZOMBIE| SIP/100-0275 BRIDGE_START | SIP/107-0274 | Local/103@100-0042;1 HANGUP | SIP/100-0275 | CHAN_END | SIP/100-0275 | HANGUP | Transfered/SIP/107-0274ZOMBIE| CHAN_END | Transfered/SIP/107-0274ZOMBIE| BRIDGE_END | SIP/107-0274 | Local/103@100-0042;1 HANGUP | Local/103@100-0042;1 | CHAN_END | Local/103@100-0042;1 | HANGUP | SIP/107-0274 | CHAN_END | SIP/107-0274 | BRIDGE_END | Local/103@100-0042;2 | SIP/103-0276 HANGUP | SIP/103-0276 | CHAN_END | SIP/103-0276 | HANGUP | Local/103@100-0042;2 | CHAN_END | Local/103@100-0042;2 | LINKEDID_END | Local/103@100-0042;2 | (33 lignes) How should these events be interpreted? Asterisk version is 11.6.0. Thanks, -BEGIN PGP SIGNATURE- iEYEARECAAYFAlKK6p0ACgkQuu7Rv+oOo/heAACeN0eMR1qwRLcdV+Tsgn9fA+6c RKcAn246hmNUU2dxivPFEziueHYRTWcS =q196 -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Redirecting a channel to Meetme fails with Hangup.
Hello List,
Good day,
We have an application, where we redirect a channel to meet me.
Sometimes the channel is getting hanged up by Asterisk, and we get an hang-up
event.
Please reply back, if any one faced such issue.
Here is the hangup event info,
-HANGUP {calleridname=unknown, connectedlinename=unknown,
uniqueid=1384413814.79523, cause=0, cause-txt=Unknown, event=Hangup,
privilege=call,all, connectedlinenum=unknown, calleridnum=03003,
channel=SIP/S0-gateway-0001363d}
Best Regards,
Ruban.S
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[asterisk-users] Communicate with barge agent
HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. Regards Akhilesh -- _ -- 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] Communicate with barge agent
On Tue, Nov 19, 2013 at 1:02 PM, akhilesh chand omakhileshch...@gmail.comwrote: HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. Regards Akhilesh Chanspy with w option - w - Enable whisper mode, so the spying channel can talk to the spied-on channel. https://wiki.asterisk.org/wiki/display/AST/Application_ChanSpy --Satish Barot satish4aster...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
