Re: [asterisk-users] Asterisk 12 CDR dst field empty - cdrlog.txt (1/1)
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Re: [asterisk-users] Asterisk 12 CDR dst field empty - cdrlog.txt (0/1)
On Wed, 15 Oct 2014 09:14:41 -0500, Matthew Jordan
mjor...@digium.com wrote:
On Wed, Oct 15, 2014 at 1:50 AM, A.Santoro n...@ecoricerche.it wrote:
Hi there,
I have installed Asterisk version 12.6 (on Debian wheezy) and I note
that, only when I make a transfer of call (attended or unattended),
the fields 'dst' and 'dcontex' in the CDR are empty.
This happen both in MySQL record and in CVS.
Someone can confirm this event?
Without more information, there's no way to tell why that would occur.
Please provide a log showing the transfer with 'cdr set debug on' enabled.
The requested log is attached.
In the last lines of the file there are the records of the call
transfer:
[Oct 17 17:52:14] VERBOSE[4314] cdr_adaptive_odbc.c: [INSERT
INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid)
VALUES ({ ts '2014-10-17 17:51:45' },'valerio
valerio','valerio','106','ebali_segreteria','SIP/valerio-','SIP/fabiana-0001','Dial','SIP/fabiana,20,rtwW',24,23,'ANSWERED',3,'1413561105.0')]
[Oct 17 17:52:15] VERBOSE[4314] cdr_adaptive_odbc.c: [INSERT
INTO cdr
(calldate,clid,src,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid)
VALUES ({ ts '2014-10-17 17:52:10' },'valerio
valerio','valerio','SIP/valerio-','SIP/francesca-0003','Dial','SIP/fabiana,20,rtwW',4,4,'ANSWERED',3,'1413561105.0')]
[Oct 17 17:52:15] VERBOSE[4314] cdr_adaptive_odbc.c: [INSERT
INTO cdr
(calldate,clid,src,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid)
VALUES ({ ts '2014-10-17 17:52:10' },'valerio
valerio','valerio','SIP/valerio-','SIP/fabiana-0002','Dial','SIP/fabiana,20,rtwW',0,0,'ANSWERED',3,'1413561105.0')]
Thanks in advance.
Eco
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[asterisk-users] Asterisk 12 CDR dst field empty
Hi there, I have installed Asterisk version 12.6 (on Debian wheezy) and I note that, only when I make a transfer of call (attended or unattended), the fields 'dst' and 'dcontex' in the CDR are empty. This happen both in MySQL record and in CVS. Someone can confirm this event? Thanks in advance. -- _ -- 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 and Patton
On Fri, 30 Apr 2010 18:52:46 +0200, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: As I said, you could think about creating 4 different SIP gateways on the Patton with 4 differing SIP ports. I don't know if the Patton will handle 4 gateways - but it might. We have 4 trunk and 4 company in our office, I was testing FOP and I would want to show the occupied trunks for inbound and outbound calls for single company. Alternatives are: - use GROUP() and GROUP_COUNT in the dialplan - use DEVICE_STATE in the dialplan This includes a lenghty example on how to monitor a BRI trunk: http://www.voip-info.org/wiki/view/Asterisk+func+device_State Philipp Hi Philipp, thanks, for your help I'll try to find a solution to use FOP. Best regards. Eco. -- _ -- 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 and Patton
Hi, we have and Asterisk server connected to a Patton Smartnode 4638 with 4 BRI. We configured 4 SIP account on Patton (1001, 1002, 1003, 1004). The system is fully functional, but we have a problem to recognize incoming calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton) or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming from SIP/1004. I have contacted Patton support, I have send configuration and debug and they told me that there is a problem of Asterisk configuration. In the sip debug on Asterisk I have seen (SIP/1001 incoming call) ... Sending to 192.168.2.122 : 5060 (no NAT) Using INVITE request as basis request - 89c9689349c54649aae566e9192c5...@192.168.2.122 Found peer '1004' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 ... (192.168.2.122 is the ip address of Smartnode.) In the Patton configuration ... gateway sip ASTERISK bind interface LAN router service default defaultserver manual 192.168.2.121 5060 loose-router registration manual 192.168.2.121 user 1001 authenticate password 36ocYTYpKxk= encrypted register display-name 1001 gateway sip ASTERISK no shutdown ... (192.168.2.121 is the ip address of Asterisk server) The call is coming from SIP/1001, but the INVITE request founds peer 1004. The problem come when I try to use FOP: I am not able to correctly connect button to trunk. Someone can help me? Thanks in advance. Eco -- _ -- 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 and Patton
On Fri, 30 Apr 2010 10:39:12 +0200, Carlo Dimaggio jaasmail...@gmail.com wrote: 2010/4/30 A.Santoro n...@ecoricerche.it Hi, we have and Asterisk server connected to a Patton Smartnode 4638 with 4 BRI. [...] Have you tried setting insecure=port,invite in the sip.conf for each sip account? Hi Carlo, thanks for your answer. Now I tried... and nothing is changed. In the following lines one the sip account of the peers (in sip.conf) [1001] username=1001 type=friend secret= dtmfmode=auto insecure=very host=dynamic port=5060 context=inbound qualify=yes disallow=all allow=ulaw allow=alaw canreinvite=no Bye. Eco -- _ -- 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 and Patton
On Fri, 30 Apr 2010 14:16:14 +0200, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton) or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming from SIP/1004. Read up on how Asterisk does user/peer matching in sip.conf on inbound calls: With all users/peers having the same IP and hostname it is the entry that was defined last in sip.conf that wins. Philipp thanks for your answer. This clears all my doubts, is not my configuration problem. why exactly do you need to know which line is in use? We have 4 trunk and 4 company in our office, I was testing FOP and I would want to show the occupied trunks for inbound and outbound calls for single company. Thanks again. Bye Eco -- _ -- 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] Skype for Asterisk
On Wed, 30 Dec 2009 11:43:59 -0800, vijay.go...@alliance-infotech.com wrote: case 2: This skype account (rexesbposolutions) has been assigned with a online virtual number (00 44 20 ). If somebody dial this number from their landline/cellphone, call is transfered to Asterisk queue but it shows some problem related to G729 codecs. following are Asterisk CLI log: I had the same problem. I contacted Digium support and this was the answer: You can download the G.729 codec from the following link: http://www.digium.com/en/docs/G729/g729-download.php Install the codec_g729a.so binary in /usr/lib/asterisk/modules/ and restart the asterisk service. I followed the advice and the problem is resolved. You can try. Bye -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Festival in Italian does not work
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