similar to: how to get call duration

This is a really rookie question: when should i use TE110P ISDN PRI Card? -- Necati DEM?R --------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100621/07d2917f/attachment.htm

Hello Sirs, > > Have you ran them through mp3check and mp3_check? > >Yep. Found a couple with a few bad frames and deleted them. Other than >that, nothing worse than a missing ID3 tag or two. Could you please give me the address of these tool(mp3check, mp3_check). I have many bad coded mp3, When Icecast streaming these bad! mp3s it kicks the connected clients. I want to delete

Dear Gluster-Users, I am experiencing RDMA problems. I have installed Ubuntu 16.04.4 running with 4.15.0-13-generic kernel, MLNX_OFED_LINUX-4.3-1.0.1.0-ubuntu16.04-x86_64 to 4 different servers. All of them has Mellanox ConnectX-4 LX dual port NICs. These four servers are connected via Mellanox SN2100 Switch. I have installed GlusterFS Server v3.10 (from Ubuntu PPA) to 3 servers. These 3

Thank you for your mail. ibv_rc_pingpong seems working between servers and client. Also udaddy, ucmatose, rping etc are working. root at gluster1:~# ibv_rc_pingpong -d mlx5_0 -g 0 ? local address:? LID 0x0000, QPN 0x0001e4, PSN 0x10090e, GID fe80::ee0d:9aff:fec0:1dc8 ? remote address: LID 0x0000, QPN 0x00014c, PSN 0x09402b, GID fe80::ee0d:9aff:fec0:1b14 8192000 bytes in 0.01 seconds =

Is infiniband itself working fine? You can run tools like ibv_rc_pingpong to find out. On Wed, Apr 25, 2018 at 12:23 PM, Necati E. SISECI <siseci at gmail.com> wrote: > Dear Gluster-Users, > > I am experiencing RDMA problems. > > I have installed Ubuntu 16.04.4 running with 4.15.0-13-generic kernel, > MLNX_OFED_LINUX-4.3-1.0.1.0-ubuntu16.04-x86_64 to 4 different servers.

Just founnd a weirdy. My end is Asterisk 1.2.32 using an IAX2 link to the US. The IP address of the remote end changed (though in the config file it's registered as a name i.e. asterisk.remote.end), my system didn't recognised the IP change, it must be cached once and then the cached value used for ever. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857

Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100802/4f58fea7/attachment.htm

Can someone tell me what I am doing wrong? Why doesn't CDR(duration) or CDR(billsec) return the correct values? cdr.conf endbeforehexten=yes extensions.conf [macro-Dial] ; ${ARG1} - Dial String exten => s,1,Dial(${ARG1},,M(post-dial)) exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long, billed for ${CDR(billsec)} seconds) The log shows: -- Executing [h

Hello List I'm moving this over from the bioC list as, although the problem I'm working on is biological, the current bottle neck is my poor understanding of R. I wonder if someone would help me with the following function. cumulMetric <- function(deMirPresGenes, deMirs){ ??? #need to match position of each miR in deMirPresGenes with its FC to form a vector of FC in correct order ?

I was just going through my call records ( stored in mysql database by cdr_MYSQL module ) and saw a record having duration = 0 and billsec of more than 50 seconds . I did a query on cdr where duration < billsec and saw that there were infact some 250 records with duration less than billsecond ( table had around 4,00,000 records) . Did anyone came across this ? I also checked csv files and they

Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l.

Hello, I am wondering if someone has got any ideas that can help solve this problem. I have a dial plan that you call into, and depending on certain conditions it calls out on a number grabbed from a database. Something like this : exten => s,n,Do something exten => s,n,Do something else exten => s,n,Dial(ZAP/g1/${OUTBOUND},${timeout}) I need to log the time the person

hi i've made some test calls, i've notice that a call of the duration of 1:29 minutes is recorded in the cdr database as 1:45 minutes, is it normal? i think that 15 seconds are too many... how can i correct this? thanks

The behaviour of my Asterisk appears to have changed suddenly without any apparent cause. The version is use is 1.4.27.1 When a call is not answered billsec is set to duration, and calls are charged. I can't see any change I could have made to cause this problem. Is it something already known in Asterisk? /voipfc

Hi I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon. With some calls, the value in the `billsec` field in the CDR is exceeding the value in the `duration` field. I didn't think this was supposed to happen? Our old installation (some ancient version, sorry not available) always had `billsec` <= `duration`. The problem does not seem to be evenly distributed

Hello guys, I'm wondering what is required and involved in order to provide a wifi/GSM handover to customers. After googling I haven't found any product/vendor. Do you have an idea ? Thanks in advance Patrick

Hi, Might be off topic but I thought it would be a good place to ask.. I am investigating switching to a hosted PBX and dumping my old Asterisk box thats been running in my office for the last few years.. The few I have found seem very expensive.. Can anyone point me to any VoIP PBX hosts in the UK? TIA -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi to all, I was looking in google and also in this mailing list, but I dont find the solution to my problem, so I subscribe me to the list in order to post this e-mail and find the solution. This is the scenario: GSM Phone ----- GSM Network ---- TDM2406E --- ASterisk 1.4.0 (*) -------- VoIP Provider ------- Sip Phone or H323 Phone The problem is that I am generating calls from SIP and also

Hello, Sorry but in the mails I have read I did not see any "shout" program comments. I have configured icecast and I am using "shout" to encode MP3s. Is the performance of shout bad or others are better then "shout"? I did not encounter any problems yet. Thank you for any comment. Tamer Demir At 12:10 15.01.2002 +0100, Maarten Stolte wrote: >Hello, >

Does anyone have any good eggdrop tcls that announce the current song or anything like that? --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages sent to