similar to: calls terminating abnormally

hi all ! we've got IAX2 protocol working between several Asterisk servers. Now we are concerned with doing bandwidth management to maintain an acceptable voice quality. We thought of prioritizing the udp traffic. ( Giving a high priority to those IAX2 udp ports.) I know that IAX2 uses udp/4569. Is there any other traffic/ports that we need to consider for bandwidth shaping w.r.t IAX2.

hi! I've got cisco 7960G working with * box. Calls could be Blind Xfered through the phone but not the supervised transfer( Message on the phone: Transfer failed). Even when I put the caller on hold and resume it later, I can't hear the other side but the otherside can hear me. (It shows as the line is connected though. Yet the respective caller entry blinks.) Any suggestions most

hi ! I have asterisk with Licensed G.729 codec enabled. Whenever I make a call using this codec a warning apears as, WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames

hi! I wanna do some arithmatic operations (addition and substraction -integer operation) inside extensions.conf. Is there a simple way to do this. If I do yy = ${xx} + 1 // say "xx" is initialized to '0' the resulting "yy" will show "0 + 1" Obiviously not the result I need. Any help !!!!! denzel.

hi We are using snome 200 IP phone with *. It works OK. But after a period of time we can't hear any sounds for any icoming or outgoing calls. I've got two of these phones. Same symptoms occur to both of these( not at the same time ) and the problem remains until the phone is completely rebooted. Don't know whether this's * or Snomes' prob. Any help would be appreciated.

hi! I'm looking for a robust hardware IP phone which supports SIP protocol inorder to implement a call centre. Have anyone used CISCO SIP phones (eg:- 7960G ) with asterisk. From what I know these CISCO IP phones are very robust and feature rich. Yet I'm nervous whether * don't like CISCO at all. Thoughts are most welcome. denzel. -------------- next part -------------- An HTML

Hi All, What the support that asterisk has to send/receive Faxes? Can I plug a FAX machine in the a FXS extension and send out Faxes? What's the codec I need to use? g.711? Also can we receive a FAX into a FXS extension in Asterisk PBX? Also I need to know if we can send/receive FSK data from/to an extension plugged into Asterisk PBX? For example if there's a phone model which can send

they disconect individually after playing one hour. Am 15.04.2020 um 19:08 schrieb Jordan Erickson: > Hi, > > Doesn't sound like anything Icecast does by default. I would inspect > your web player / browser for clues. Do all clients disconnect at the > same time every hour, or do they individually disconnect after playing > for an hour each? > > > Cheers, >

sry i had forgot to answer at the list. there is no restart at the logfiles of icecast, and there are no listener behind Am 15.04.2020 um 16:40 schrieb tonton Th: > On 4/15/20 4:06 PM, Railgun wrote: >> Hi I have a problem, the webplayer, Oder other clients (only VLC not) >> get >> most time disconect after exactly one hour. >> >> Somone can help me solv this

Perhaps check the mount settings in the icecast config file: <mount type="normal"> <mount-name>/example-complex.ogg</mount-name> <username>othersource</username> <password>hackmemore</password> <max-listeners>1</max-listeners>* <max-listener-duration>3600</max-listener-duration>* -

Hi I have a problem, the webplayer, Oder other clients (only VLC not) get most time disconect after exactly one hour. Somone can help me solv this problem? I using icecast 2.4.4 on Ubuntu 18.04 And ezstream as source. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/icecast/attachments/20200415/7aa4270b/attachment.htm>

Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to "help" Asterisk notice the disconect?

Dear All I have jus migrated to centos 5 and jus wondering why n where this problem could be server A which is running for the past 2 years used as my primary dns and mail server redhat linux 8 sendmail ver-8.12.5-7 bind ver -9.2.1-9 since the server is old and is prone to all security problems i have a new machine which i have installed the following and will replace my existing redhat dns +

Dear All I have jus migrated to centos 5 and jus wondering why n where this problem could be server A which is running for the past 2 years used as my primary dns and mail server redhat linux 8 sendmail ver-8.12.5-7 bind ver -9.2.1-9 since the server is old and is prone to all security problems i have a new machine which i have installed the following and will replace my existing

Hi... I've have problem with my file and mail server. I have a Centos 4 with file and mail server for my office network, which is compose with Linux (xandros 4) and windows (2k) work stations. The problem is that suddenly some linux machines can't access the the remote files on the server, but they can see the server in the file manager, but there is no files. When I enter as

Hi, I have problems with pop3 migration, the server disconect the dovesync when fetching mails after 5 minutes with this error in server log: xx.xx.xx.xx [28F4] 11:54:05 <<< RETR 6548 xx.xx.xx.xx [28F4] 11:54:05 >>> +OK 47522 octets xx.xx.xx.xx [28F4] 11:54:05 <<< RETR 6549 xx.xx.xx.xx [28F4] 11:54:05 >>> +OK 355289 octets xx.xx.xx.xx [28F4] 11:54:05

hi! I'm trying to get zapras working in GSM csd network. Whenever a dialup call is initiated from the mobile to the * gateway the following appears in the log and zapras terminates. Phone gives the error dialup not answered. ==> /var/log/messages <== pppd[2310]: Plugin zaptel.so loaded. pppd[2310]: Zaptel Plugin Initialized pppd[2310]: Using zaptel device 'stdin' pppd[2310]:

Can a worker kill themselves when they''re ''done''? Or do I have to do that either from the controller or the worker manager? Thanks, Bill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://rubyforge.org/pipermail/backgroundrb-devel/attachments/20061023/11dd429c/attachment.html

have a look in the encoder config file [ not icecast] and see if there is a time limit there. in Icecast check you have plenty of capacity <limits>         <clients>100</clients>         <sources>20</sources>         <queue-size>524288</queue-size>         <client-timeout>30</client-timeout>        

<P>hi,</P> <P>Can anybody pls tell me how to activate loggin CDR on mysql db. I tried&nbsp;editing the /etc/asterisk/cdr_mysql.conf file and recompiling asterisk, it didn't work. Normal loggin on Master.csv file works fine.</P> <P>Thank you inadvance,</P> <P>Surajee</P> <P>&nbsp;</P><br> --------------This mail sent