Re: [asterisk-users] users can not hear the audio playback sometimes
Maybe there is a way you can define a custom trunk and basically copy the standard part of the dialplan but with the extra line you added? Sorry I dont know the product so cant tell you how to do it. No Problem, it was easy to fix. There is a file which is ignored by freepbx while doing the reload, so i copied my trunk into that file. thanks shantanu On 25/09/13 13:57, Kumar Shantanu wrote: Thank you Gareth, It worked like a charm. The only problem I am having is now, when I do some changes in my freepbx and reload it just rewrites my dial play , I will try to fix it though. Thanks again I did see in the console output it doing a GotoIf() and checking if a custom trunk was defined just before the original dial command. Maybe there is a way you can define a custom trunk and basically copy the standard part of the dialplan but with the extra line you added? Sorry I dont know the product so cant tell you how to do 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 -- _ -- 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] Please check Important..
This is a phishing URL. Dont click it. Moderators: can you put this lovely person on the moderated list? On 9/25/13 21:11:44, Code Lover wrote: Hi, I have attached an important document via Google docs, Please check the link below for additional security you will be required to sign in with your email before viewing / downloading the document. *removed* -- Thank You, Abdul Lateef Senior – Development Barwa Bank Doha Qatar --- Please do not print this e-mail unless it is absolutely necessary. -- _ -- 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] mysql CDRs in web based tool
but i do not know how to interface the CDRs. has anyone used this tool or any other similar tool? how about something like this: pbx@pbx:~$ grep -v ^; /etc/asterisk/cdr_mysql.conf [global] hostname=localhost dbname=dbname table=tablename password=password user=username port=3306 sock=/tmp/mysql.sock timezone=CET ; Previously called usegmtime [columns] alias start = calldate pbx@pbx:~$ -- _ -- 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] Queue Management
Dear All, I have six different campaign and 5 different agent have login on that campaign.*Same thing i have done using agi and database,i never use queue management on this scenario. Agent** can also shuffling one campaign to anther campaign. * Now i want to do some work with queue.I want to use single queue to managing this. Eg: campaign Agent Login A a_1,a_3 (In campaign A 2 agents are login) B a_2,a_1 (In campaign B 2 agents are login) C a_3,a_1,a_4 (In campaign C 3 agents are login) D a_4,a_5,a_3 (In campaign D 3 agents are login) E a_1,a_3,1_2 (In campaign E 3 agents are login) F a_5,a_4(In campaign F 2 agents are login) When a call come to campaign A that call goes to agent a_1 or a_3 not goes to other campaigns agents. 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] Generating a different countries ringtone on a per call basis
On Wed, Sep 25, 2013 at 6:45 AM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: We can use the Dial() command with the 'r' option in order to generate the UK ringtone (as we are UK based the default is UK). How do we generate a USA ringtone for example? I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us' (and calling Progress() beforehand) and although this works for Playtones() the Dial command still continues to play the UK ringtone. Any ideas? Try the following: extension = 6001,1,Set(CHANNEL(tonezone)=us) same = n,Dial(SIP/6001,,r(ring)) The argument passed to the r option should be the specific tone in the category of the tonezone you are setting. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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 / SIP-Call / AGI-Script / SIGHUP after Answer
Hi, I am facing a (for me) strange problem. When placing a SIP-Call I normally get connected and the dialplan is executed. The Call-Flow is controlled by a PHP-Agi-Script. The script answers the call (via AGI-Command) and a simple voicefile is played. SOMETIMES(!) I get disconnected immediately after the Answer - without any reason. This happens about all fifth call. Later on you will find my SIP-Debug-Output. I can see a BYE-Message. But why? AGI-Debug-Messages: (yes - I can the result is -1 but why? Normally it is 0) -- snip -- SIP/thorsten-01f8AGI Rx Answer SIP/thorsten-01f8AGI Tx 200 result=-1 -- snip -- SIP-Debug-Messages: -- snip -- --- SIP read from UDP:217.92.105.86:51861 --- INVITE sip:3...@myhost.org SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:51861;rport;branch=z9hG4bKPj47b1a62ac3744acd996426618d90388f Max-Forwards: 70 From: Thorsten (myhost) sip:thors...@myhost.org;tag=4313e82f4af9423bab056113e5e05713 To: sip:3...@myhost.org Contact: sip:03794281@192.168.1.2:51861 Call-ID: a19e81e8a2d74f718e1263ab3fd3b328 CSeq: 28484 INVITE Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: 100rel, replaces, norefersub, gruu User-Agent: Blink 0.5.0 (Windows) Content-Type: application/sdp Content-Length: 386 v=0 o=- 3589198761 3589198761 IN IP4 192.168.1.2 s=Blink 0.5.0 (Windows) c=IN IP4 192.168.1.2 t=0 0 m=audio 10054 RTP/AVP 108 99 98 9 0 8 96 c=IN IP4 192.168.1.2 a=rtcp:10055 a=rtpmap:108 opus/48000 a=rtpmap:99 speex/32000 a=rtpmap:98 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv - --- (13 headers 17 lines) --- Sending to 217.92.105.86:51861 (no NAT) Sending to 217.92.105.86:51861 (no NAT) Using INVITE request as basis request - a19e81e8a2d74f718e1263ab3fd3b328 Found peer 'thorsten' for 'thorsten' from 217.92.105.86:51861 --- Reliably Transmitting (NAT) to 217.92.105.86:51861 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:51861;branch=z9hG4bKPj47b1a62ac3744acd996426618d90388f;received=217.92.105.86;rport=51861 From: Thorsten (myhost) sip:thors...@myhost.org;tag=4313e82f4af9423bab056113e5e05713 To: sip:3...@myhost.org;tag=as7b1fc32b Call-ID: a19e81e8a2d74f718e1263ab3fd3b328 CSeq: 28484 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=myhost, nonce=0d688867 Content-Length: 0 Scheduling destruction of SIP dialog 'a19e81e8a2d74f718e1263ab3fd3b328' in 32000 ms (Method: INVITE) --- SIP read from UDP:217.92.105.86:51861 --- ACK sip:3...@myhost.org SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:51861;rport;branch=z9hG4bKPj47b1a62ac3744acd996426618d90388f Max-Forwards: 70 From: Thorsten (myhost) sip:thors...@myhost.org;tag=4313e82f4af9423bab056113e5e05713 To: sip:3...@myhost.org;tag=as7b1fc32b Call-ID: a19e81e8a2d74f718e1263ab3fd3b328 CSeq: 28484 ACK User-Agent: Blink 0.5.0 (Windows) Content-Length: 0 - --- (9 headers 0 lines) --- --- SIP read from UDP:217.92.105.86:51861 --- INVITE sip:3...@myhost.org SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:51861;rport;branch=z9hG4bKPj71edb9caa0e84a52b14777e7d949bc2a Max-Forwards: 70 From: Thorsten (myhost) sip:thors...@myhost.org;tag=4313e82f4af9423bab056113e5e05713 To: sip:3...@myhost.org Contact: sip:03794281@192.168.1.2:51861 Call-ID: a19e81e8a2d74f718e1263ab3fd3b328 CSeq: 28485 INVITE Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: 100rel, replaces, norefersub, gruu User-Agent: Blink 0.5.0 (Windows) Authorization: Digest username=thorsten, realm=myhost, nonce=0d688867, uri=sip:3...@myhost.org, response=c1a2ab209d255b4ee805edd4de48380a, algorithm=MD5 Content-Type: application/sdp Content-Length: 386 v=0 o=- 3589198761 3589198761 IN IP4 192.168.1.2 s=Blink 0.5.0 (Windows) c=IN IP4 192.168.1.2 t=0 0 m=audio 10054 RTP/AVP 108 99 98 9 0 8 96 c=IN IP4 192.168.1.2 a=rtcp:10055 a=rtpmap:108 opus/48000 a=rtpmap:99 speex/32000 a=rtpmap:98 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv - --- (14 headers 17 lines) --- Sending to 217.92.105.86:51861 (NAT) Using INVITE request as basis request - a19e81e8a2d74f718e1263ab3fd3b328 Found peer 'thorsten' for 'thorsten' from 217.92.105.86:51861 == Using SIP RTP CoS mark 5 Found RTP audio format 108 Found RTP audio format 99 Found RTP audio format 98 Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 96 Found unknown media description format opus for ID 108 Found audio description format speex for ID 99 Found audio description format speex for ID 98 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format
[asterisk-users] Problems sending log to rsyslog
So I have asterisk 1.8.23 and want to send my logs to rsyslog. I tell asterisk to use syslog in addition to messages: root@voip:~# tail -10 /etc/asterisk/logger.conf ;debug = debug console = notice,warning,error ;console = notice,warning,error,debug messages = notice,warning,error ;full = notice,warning,error,debug,verbose,dtmf,fax ;syslog keyword : This special keyword logs to syslog facility ; syslog.local0 = notice,warning,error ; root@voip:~# After reloading (asterisk -rx 'logger reload') the logger, it seems that Asterisk is happy: root@voip:~# asterisk -rx 'logger show channels' Channel Type StatusConfiguration --- --- syslog.local0 Syslog Enabled- NOTICE WARNING ERROR /var/log/asterisk/messages File Enabled- NOTICE WARNING ERROR Console Enabled- NOTICE WARNING ERROR root@voip:~# So I set rsyslog: root@voip:~# fgrep asterisk /etc/rsyslog.d/50-default.conf local0.* /var/log/asterisk/messages.log root@voip:~# and restart it. And then check the asterisk log directory: root@voip:~# ls -lh /var/log/asterisk/ total 3.7M drwxr-xr-x 2 asterisk asterisk 4.0K Jul 22 20:57 cdr-csv drwxr-xr-x 2 asterisk asterisk 4.0K Jun 28 14:16 cdr-custom -rw-rw 1 asterisk asterisk 252K Sep 26 09:37 messages -rw-rw 1 asterisk asterisk 248K Sep 22 05:14 messages.1 -rw-r- 1 syslog adm 0 Sep 26 06:47 messages.log -rw-rw 1 asterisk asterisk 118 Sep 26 10:07 queue_log root@voip:~# It does not seem like much is being written to messages.log compared to messages. Anything I missed? -- _ -- 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] Generating a different countries ringtone on a per call basis
On 26/09/13 14:59, Rusty Newton wrote: On Wed, Sep 25, 2013 at 6:45 AM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: We can use the Dial() command with the 'r' option in order to generate the UK ringtone (as we are UK based the default is UK). How do we generate a USA ringtone for example? I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us' (and calling Progress() beforehand) and although this works for Playtones() the Dial command still continues to play the UK ringtone. Any ideas? Try the following: extension = 6001,1,Set(CHANNEL(tonezone)=us) same = n,Dial(SIP/6001,,r(ring)) The argument passed to the r option should be the specific tone in the category of the tonezone you are setting. Thanks. I did try that as pretty much the first thing I tried but it continued to play the UK ring tone. Its not a big issue as we can work around it by playing music on hold instead which is a recording of the required ring tone. Having asterisk generate it just seemed the neater option. -- _ -- 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] Problems sending log to rsyslog
On 26/09/13 15:25, Mauricio Tavares wrote: So I have asterisk 1.8.23 and want to send my logs to rsyslog. I tell asterisk to use syslog in addition to messages: root@voip:~# tail -10 /etc/asterisk/logger.conf ;debug = debug console = notice,warning,error ;console = notice,warning,error,debug messages = notice,warning,error ;full = notice,warning,error,debug,verbose,dtmf,fax ;syslog keyword : This special keyword logs to syslog facility ; syslog.local0 = notice,warning,error ; root@voip:~# After reloading (asterisk -rx 'logger reload') the logger, it seems that Asterisk is happy: root@voip:~# asterisk -rx 'logger show channels' Channel Type StatusConfiguration --- --- syslog.local0 Syslog Enabled- NOTICE WARNING ERROR /var/log/asterisk/messages File Enabled- NOTICE WARNING ERROR Console Enabled- NOTICE WARNING ERROR root@voip:~# So I set rsyslog: root@voip:~# fgrep asterisk /etc/rsyslog.d/50-default.conf local0.* /var/log/asterisk/messages.log root@voip:~# and restart it. And then check the asterisk log directory: root@voip:~# ls -lh /var/log/asterisk/ total 3.7M drwxr-xr-x 2 asterisk asterisk 4.0K Jul 22 20:57 cdr-csv drwxr-xr-x 2 asterisk asterisk 4.0K Jun 28 14:16 cdr-custom -rw-rw 1 asterisk asterisk 252K Sep 26 09:37 messages -rw-rw 1 asterisk asterisk 248K Sep 22 05:14 messages.1 -rw-r- 1 syslog adm 0 Sep 26 06:47 messages.log -rw-rw 1 asterisk asterisk 118 Sep 26 10:07 queue_log root@voip:~# It does not seem like much is being written to messages.log compared to messages. Anything I missed? Have you checked the /var/log/asterisk directory permissions? I dont know how rsyslog is setup on your system but its possible it gets started as root, sees the destination file doesnt exist so creates it and sets the file permissions, and then drops down to running as the syslog user. At this point it doesnt have write permission to the /var/log/asterisk directory so cannot append to the file. -- _ -- 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] Generating a different countries ringtone on a per call basis
On Thu, Sep 26, 2013 at 10:08 AM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 26/09/13 14:59, Rusty Newton wrote: Try the following: extension = 6001,1,Set(CHANNEL(tonezone)=us) same = n,Dial(SIP/6001,,r(ring)) The argument passed to the r option should be the specific tone in the category of the tonezone you are setting. Thanks. I did try that as pretty much the first thing I tried but it continued to play the UK ring tone. Its not a big issue as we can work around it by playing music on hold instead which is a recording of the required ring tone. Having asterisk generate it just seemed the neater option. Are you sure you specified an argument to the 'r' option? Or did you just try 'r' without an argument? For me.. if I specify a uk tonezone, to get it playing uk tones I have to specify an argument to the 'r' option. If I try just 'r' by itself then I get US tones. You would think, that without specifying an argument, it should default to the tonezone in use on the channel. That may be a bug or oversight. What version of Asterisk were you using, and what channel type? -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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] mysql CDRs in web based tool
Στις 26/9/2013 2:48 μμ, ο/η ad...@3a.hu έγραψε: but i do not know how to interface the CDRs. has anyone used this tool or any other similar tool? how about something like this: pbx@pbx:~$ grep -v ^; /etc/asterisk/cdr_mysql.conf [global] hostname=localhost dbname=dbname table=tablename password=password user=username port=3306 sock=/tmp/mysql.sock timezone=CET ; Previously called usegmtime [columns] alias start = calldate pbx@pbx:~$ It seems nice, but ho do u use 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
