Re: [asterisk-users] click2call for conferencing two mobile numbers
Thanks Stiles Trying as u asked to do Regards On Fri, May 6, 2016 at 6:10 PM, A J Stiles <asterisk_l...@earthshod.co.uk> wrote: > On Friday 06 May 2016, Alok Srivastava wrote: > > Dear List > > wanna configure click2call in such a manner that my asterisk box call two > > mobile numbers and connect both numbers to talk. I have configured voip > > gateway, my internal and external calls are working fine. > > please help , > > You ought to be able to do this just using call files. > > All you have to do is inject a callfile (format is explained on the Wiki) > into the folder /var/spool/asterisk/outgoing/ . You have to do this > within a > CGI script, so you can pass the two end numbers to that script when the > button > is clicked. > > Note that depending on the block size used on the underlying device, you > probably should first create the file in some temporary location and then > mv it > to ...outgoing/ . Otherwise there is a danger of Asterisk reading an > incomplete file and doing nothing. Only if you know the entire file is > definitely going to be smaller than one block, can you get away with > creating > it in place. > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot uk . > > -- > _ > -- 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] click2call for conferencing two mobile numbers
Dear List wanna configure click2call in such a manner that my asterisk box call two mobile numbers and connect both numbers to talk. I have configured voip gateway, my internal and external calls are working fine. please help , abhi -- _ -- 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] NOT able to call on local extensions while successfully call on external mobile no.(using SONETEL account)
*Dear List*
Plz help, i am not much experienced with asterisk. i configured it on
ubuntu 12.04. no problem when i call any mobile no(0091XX) but when
i call on my local asterisk no.(101,102 or 105) it is not connecting
giving error
Auto fallthrough, channel 'SIP/lucknow-006f' status is 'CHANUNAVAIL'
*while when i call 200 it is playing audiofile successfully. Please help *here
is my sip.conf and extensions.conf.
thanks.
*=sip.conf*
[general]
context=unauthenticated
allowguest=yes
srvlookup=yes
udpbindaddr=0.0.0.0
tcpenable=no
register = supp...@mydomain.net:passw...@sip.sonetel.com
outboundproxy=sip.sonetel.com
[usa_number]
type=friend
dtmfmode=rfc2833
context=hello123
host=sip.sonetel.com
username=support
secret=password
nat=yes
fromdomain=mydomain.net
outboundproxy=sip.sonetel.com
insecure=invite
disallow=All
allow=alaw
allow=ulaw
allow=gsm
[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=yes
secret=s3CuR#p@s5
dtmfmode=auto
disallow=all
allow=ulaw
; define a device name and use the office-phone template
[bombay](office-phone)
; define another device name using the same template
[lucknow](office-phone)
[test5](office-phone)
===
*extensions.conf===*
[LocalSets]
exten = _00X.,1, Answer
exten = _00X.,n, Set(CALLERID(num)=support)
exten = _00X.,n, Dial(SIP/${EXTEN}@support)
exten = _00X.,n, Hangup
exten = 101,1,Dial(SIP/lucknow)
exten = 102,1,Dial(SIP/bombay)
exten = 105,1,Dial(SIP/test5)
exten = 200,1,Answer()
same = n,Playback(an_audio_file)
same = n,Hangup()
[hello123]
exten = support,1,Answer
exten = support,n,Playback(Enterprise-Welcome-message);
exten = support,n, Hangup
===
regards
abhi
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Re: [asterisk-users] How to use phpmyadmin to remotely access asterisk mysql database?
yes u can access form same phpmyadmin both database, depends, for which database u entered userid and password on phpmyadmin login page. On Thu, Sep 11, 2014 at 2:06 PM, rafa alfurqan rafa.alfur...@gmail.com wrote: Hi, thank you for your repplied, As you're on Ubuntu, you can begin with $ sudo apt-get install phpmyadmin i did that, so what i have to do for the configuration in asterisk so i could remote to asterisk database from phpmyadmin? Also, 10.04 is a really old Ubuntu release now, even although it is a Long Term Support one. Consider upgrading to 14.04. You can apt-get dist-upgrade straight from an LTS release to the next LTS release, without needing to go through all the intermediate releases. really appreciate for the advice, i'll do that after i could remote to asterisk database from phpmyadmin. actually i have installed freeradius-server on my ubuntu too, and i could remote the database freeradius from phpmyadmin. is it possible if same phpmyadmin could remote database from freeradius-server and asterisk (they are on same server)? thank you -- _ -- 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] res_calendar.so and res_calendar_caldav.so
dear lists trying to integrate google calendar with asterisk 1.8.20.1. but 'calendar show calendars' not showing anything. when i run 'show modules' on asterisk prompt. it is not showing res_calendar_caldav.so module, only showing res_calendar.so module . is there anything wrong with google API. plz help. abhi -- _ -- 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] testing asterisk11 on single machine
can i test my asterisk11 on a single machine on which asterisk is installed that sounds are working from both end properly. i have installed asterisk 11 on ubuntu12.04 with twinkle soft phone. regards abhi -- _ -- 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] click to call
dear is there any study material for implementing click to call in asterisk. plz help. thanks regards -- _ -- 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] sip set debug on always showing error
dear please Help. I am continously getting this message after sip set debug on. and not getting clear voice from both side. --- Transmitting (NAT) to 122.163.193.94:1893 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106:5060 ;branch=z9hG4bK-323331-c72a5bb7c840f6a183902fdcca79241b;received=122.163.193.94;rport=1893 From: 2002 sip:2002@122.160.154.189;tag=5a1cc54c To: 2002 sip:2002@122.160.154.189;tag=as64f1f102 Call-ID: 8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0 CSeq: 245 OPTIONS Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 Scheduling destruction of SIP dialog '8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '6636e2ee56cce3b0e2287ebe51962c84@0:0:0:0:0:0:0:0' Method: OPTIONS Really destroying SIP dialog '4ec2032c2b9a58e1cf2d1afa70c1970b@0:0:0:0:0:0:0:0' Method: OPTIONS -- _ -- 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] sip set debug on always showing error
previously i was using for codec allow=all after that i changed disallow=all allow=silk24 and i also change softph x-lite from jitsi(because of codec) now voice was coming fine from both side. But when i came to home from office not getting voice from both side. Threr is Airtel Broadband at my place. On Thu, Jul 5, 2012 at 3:36 PM, SamyGo govoi...@gmail.com wrote: Hi, *CSeq: 245 OPTIONS * * * This is just SIP keep-alive. It has nothing to do with any Call-media degradation. If you are not getting clear voice check the codecs, network latency/delay/loss/jitter parameters. BR Sammy On Thu, Jul 5, 2012 at 2:34 PM, alok srivastava alok...@gmail.com wrote: dear please Help. I am continously getting this message after sip set debug on. and not getting clear voice from both side. --- Transmitting (NAT) to 122.163.193.94:1893 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106:5060 ;branch=z9hG4bK-323331-c72a5bb7c840f6a183902fdcca79241b;received=122.163.193.94;rport=1893 From: 2002 sip:2002@122.160.154.189;tag=5a1cc54c To: 2002 sip:2002@122.160.154.189;tag=as64f1f102 Call-ID: 8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0 CSeq: 245 OPTIONS Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 Scheduling destruction of SIP dialog '8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '6636e2ee56cce3b0e2287ebe51962c84@0:0:0:0:0:0:0:0' Method: OPTIONS Really destroying SIP dialog '4ec2032c2b9a58e1cf2d1afa70c1970b@0:0:0:0:0:0:0:0' Method: OPTIONS -- _ -- 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 -- _ -- 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] port 5060 is blocked by ISP
REGISTER Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: sip:9000@192.168.6.25;expires=3600 Date: Wed, 04 Jul 2012 14:08:17 GMT Content-Length: 0 Scheduling destruction of SIP dialog 'qajoimvsihxpsff@alok-Inspiron-N5110' in 32000 ms (Method: REGISTER) --- SIP read from UDP:122.163.193.94:1801 --- - Really destroying SIP dialog 'qajoimvsihxpsff@alok-Inspiron-N5110' Method: REGISTER regards abhi On Mon, Jul 2, 2012 at 5:22 PM, SamyGo govoi...@gmail.com wrote: actually its a one-way audio issue due to NAT ! alok , please explain your network flow for end to end client-server-client. You may need to set nat=yes for your sip peer behind NAT. If the server is behind NAT router/firewall use externip=public.ip.of.server field. Also provide sip traces of this call. Another thing to do for your learning. Execute wireshark on both softphone systems and set sip | rtp as filter and see where are the RTP streams going on each end ! Take a complete capture on Asterisk server by executing the command sip set debug on and make a call. BR Sammy On Mon, Jul 2, 2012 at 4:39 PM, Thomas Kenyon dig...@sanguinarius.co.ukwrote: alok srivastava wrote: dear i have configured properly asterisk. At the one end i am using x-lite soft ph and another end twinkle. call is going properly from both end but after picking the phone not able to listen other one. when i checked the port 5060 on the asterisk server it is always showing closed while i have flushed all the rules from iptables (iptables -F) PORT STATE SERVICE VERSION 5060/tcp closed sip telnet localhost 5060 (could not connect) regards alok SIP is only used to setup (and stop etc.) the call. The actual audio is sent via rtp. /etc/asterisk/rtp.conf Should tell which ports asterisk is using for rtp, you will need to make sure that the remote host can connect to these ports. There are lots of articles around on how to resolve this. -- __**__**_ -- 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-**usershttp://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 -- _ -- 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] port 5060 is blocked by ISP
dear i have configured properly asterisk. At the one end i am using x-lite soft ph and another end twinkle. call is going properly from both end but after picking the phone not able to listen other one. when i checked the port 5060 on the asterisk server it is always showing closed while i have flushed all the rules from iptables (iptables -F) PORT STATE SERVICE VERSION 5060/tcp closed sip telnet localhost 5060 (could not connect) regards alok -- _ -- 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
