similar to: PSTN to SMS and SMS to PSTN

I have P4 2.50GHz RAM 4GB, Asterisk how many calls handle using H.323 to SIP conversion on this server? Regards, --------------------------- Muhammad Asif Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090210/6d5cb26b/attachment.htm

Dear All, I want set call limit for IAX2 users at the time incoming and outgoing, Please help me how i can set call limit as asterisk support for SIP users. -------------------------- Thanks & Regards, M. Asif Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101011/1286342e/attachment.htm

Hi, I am facing some issues while using MixMonitor and StopMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b) exten => s,2,Dial(SIP/101,13) exten => s,3,StopMonitor() exten => s,4,NoOp(Dial Status: ${DIALSTATUS}) exten => s,5,Goto(sss-${DIALSTATUS},1) exten => sss-NOANSWER,1,VoiceMail(777 at salesvoice) exten =>

Hi Sir I am very new user of R for the research project on multilevel logistic regression. There is confusion about bugs() function in R and BUGS software. Is there any relation between these two? Is there any comprehensive package for Multilevel Logistic modelling in R? Please guide in this regard. Thank You RAZA --------------------------------- Boardwalk for

Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-00000200 prevented Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am running asterisk 1.8.14.0, It was running fine for last few days and suddenly crashed today In logs I can see that abrt tried to save the core dump but it couldn't Mar 6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac ip 0000000000533c19 sp 00007f7db9ce3af0 error 4 in asterisk[400000+1d1000] Mar 6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528

Hi Friends, We have created a video based training for Asterisk in English and Urdu. Please check them and let us know how we can improve them for no-voice users. http://www.youtube.com/watch?v=KXq9g8UiGnQ http://www.youtube.com/watch?v=MID2RvgdD7s http://www.youtube.com/watch?v=_LbDUdAGfSY http://www.youtube.com/watch?v=J9Chkrg7E-M http://www.youtube.com/watch?v=MsC12wc9ZnU

Hi all I have been googling and wiki-ing and have found a number of potential solutions to my questions, but I don't want to have to play about for too long and risk messing up my * box now I've just got it working, if one of you kind folk could offer your 2 penneth, (being a Brit I'll have none of this cents business ;] ). I want to send an SMS message whenever I get a voicemail

Hello I am working on TDM2400p. I am having some problems like: when i connect my analog phone with the card there is no dial tone, but i can dial any extension... but after that i can't hear any voice from my receiver i have used different phone sets but still i cant communicate with other extension. Please help me out. Thank you Regards Ali Raza -------------- next part -------------- An

Hi does anyone have a definitive list of countries other than those listed on the wiki which are supporting app_SMS on landlines using ETSI ES 201 912 ?? Thanks Tim Robinson Basingstoke, UK

Hi, I've tested 1.8.6.0, 1.8.4.0 and 1.8.0 I can get proper start and answer time but not the end time of call <SIP/11-00000000>AGI Rx << GET VARIABLE CDR(start) <SIP/11-00000000>AGI Tx >> 200 result=1 (2011-12-16 18:34:48) <SIP/11-00000000>AGI Rx << GET VARIABLE CDR(end) <SIP/11-00000000>AGI Tx >> 200 result=1 (2011 12-16 18:34:48)

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default] exten=>_.,1,Dial(IAX2/${EXTEN})

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default]

Hi List, Which will be the appropriate variable to get called peer IP address? I tried following channel variables peerip, recvip, URI, from and following SIP channel variables: SIPURI,SIPDOMAIN They all return calling peer IP but not the destination/called peer IP. unfortunately set(CDR(calledip)=${CHANNEL(to)}) doesn't work Regards, Zohair Raza -------------- next part --------------

The Asterisk Development Team has announced the release of Asterisk 1.4.29. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.29 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix to Monitor which previously assumed the file to write to

The Asterisk Development Team has announced the release of Asterisk 1.4.29. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.29 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix to Monitor which previously assumed the file to write to

The Asterisk Development Team has announced the release of Asterisk 1.6.0.21. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.21 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix to Monitor which previously assumed the file to

The Asterisk Development Team has announced the release of Asterisk 1.6.0.21. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.21 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix to Monitor which previously assumed the file to

Hi, I have Asterisk 1.4.4 on my linux box. Whenever i try to kick a participant in conference say "59681446" using following command meetme kick 59681446 1 where "1" is the participant number, following are the actions that asterisk takes * IVR "You have been kicked from this conference" is played. * Participant is taken out from that conference

Respected Sir, I want your help regarding an issue on asterisk. I hope my mail will not disturb your daily routine. My issue is I am connecting two asterisk over IIAX2/SIP trunk. I have successfully connected multiple server and every client from one server to call any other server's client. But problem is I want to use Speex@ 2.15kbs and also packetization time is 60ms but I can not