search for: surajee

hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030906/ed1d46cf/attachment.htm

...I E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee --------------This mail sent through OmniBIS.com--------------

...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 through OmniBIS.com--------------

...;BR>dchan=16</P> <P>zapata.conf,</P> <P>;E100p card<BR>switchtype=EuroISDN<BR>signalling=pri_cpe<BR>pridialplan=unknown<BR>context=incoming<BR>group = 2<BR>channel => 1-15,17-31</P> <P>Thanks inadvance,</P> <P>Surajee</P><br> --------------This mail sent through OmniBIS.com--------------

...been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great performance) Thank you very much, Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030601/01aa5a49/attachment.htm

...ot (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome this problem, if yes, please let me/us know.</P> <P>Thank you inadvance,</P> <P>Surajee</P><br> --------------This mail sent through OmniBIS.com--------------

hi All, quit new to asterisk, can anybody tell me whether Wildcard E400P and Wildcard E100P support R2 CAS protocol. if they do, what is the value, i should set to 'signalling' parameter in the zapata.conf file? Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030519/fcc55987/attachment.htm

...ocess 2 frames WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames these warning messages come a lot, but still u can have a normal voice conversation. but this warning messages are very irritative.. does anybody has an idea on this? Thanks inadvance, Surajee --------------This mail sent through OmniBIS.com--------------

...DBGet, Can i put about 1000 keys in a single family, (only once for the lifetime) for ex. exten => _X.,5,DBput(family/key1=${val}) ... exten => _X.,5,DBput(family/key1000=${val}) like above and if i later retrieve it, randomely, with inbound calls, will it affect performance? Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030909/dc9b19d3/attachment.htm

...runk10@trunk50]/1 -- Hungup 'Zap/2-1' the above, "Zap/1-1 answered IAX2[trunk10@trunk50]/1" line comes as soon as that Zap/1-1 line starts ringing, while the Zap/2-1 is hungup. in our zapata conf, 'callprogress=yes' is commented out. any idea why is this happening? Surajee --------------This mail sent through OmniBIS.com--------------

.../FONT></DIV> <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV> <DIV><FONT face=Arial size=2>Thanx inadvance,</FONT></DIV> <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV> <DIV><FONT face=Arial size=2>Surajee</FONT></DIV><br> --------------This mail sent through OmniBIS.com--------------

...onnects the line. Can anybody give us a solution for this. In the near future, we are going to add some E1 lines too(with E400P cards), once this is done, will the above call disconnection problem occur in that configuration too..or is this a common problem only with analog ? Thank you very much, Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030530/8d571f13/attachment.htm

...too. But it seems like SJPhone is using a version of G.729 other than Codec G.729 (Annex B) , probably 'Annex A'. What we want is to have a smoother communication when we use a SIP phone on the Dialup connection. Can anybody come up with a good idea for this problems, Thank you very much, Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030514/ded057e9/attachment.htm

hi, can we have arrays in contexts? i tried like this, but didn't work :-( declaration myarray[0]=192.168.3.4 myarray[1]=192.168.3.1 usage myvalue = ${myarray[${myval}]} pls tell a way to do this Thanx a lot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030521/f8b61c89/attachment.htm

...been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great performance) Thank you very much, Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030530/3eacf366/attachment.htm

...the speech circuit (or channel) has an associated E-wire and M-wire for signalling. </P> <P>Can anybody pls tell me as to which type of lines should i get from the analog pbx,<BR>whether normal analog lines or E&M lines?</P> <P>Thanx in advance,</P> <P>Surajee</P><br> --------------This mail sent through OmniBIS.com--------------

...following, but it doesn't seems to be helping when it comes to call transfering ... exten => s,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,5,ResponseTimeout,10 ; Set Response Timeout to 10 seconds ... can anybody gv me an idea? Thank you inadvance, Surajee --------------This mail sent through OmniBIS.com--------------

hi, hi we have G729 codec from Digium, without the G729 codec, we can do the hash transfers to other sip phones fine. but once we are using the G729 codec, the asterisk is not responding to hash transfer, ie, when we press "#" it does not detect it and says "transfer..", is this a problem with G729 codec? (for testing purposes we have bought licenses for 2 chs) this also

...> > Inband DTMF detection works only on G.711 frames. > It seems that the codec negotiation phase ended up > with the GSM codec, so you get these messages. > A quick fix would be to comment out this informative > line in 'dsp.c' file. > > > Michael. > > > surajee@infotechs.lk wrote: > > hi, > > > > i have connected a SNOM 200 to the asterisk. here are my settings, > > > > Codecs > > ------- > > > > Default codec - g.711u/g.711a > > Packet size - 20ms > > Negotiation - Interoperable > > Type -...

...following, but it doesn't seems to be helping when it comes to call transfering ... exten => s,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,5,ResponseTimeout,10 ; Set Response Timeout to 10 seconds ... can anybody gv me an idea? Thank you inadvance, Surajee --------------This mail sent through OmniBIS.com--------------