similar to: Help Required (How to acheive packetization time of 60ms over SIP/IAX2 trunk)

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 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

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

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

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 all, I need to be able to adjust packet sizes and found the patch at http://bugs.digium.com/view.php?id=5162 Thus, I checked out and compiled http://svn.digium.com/view/asterisk-old/team/group/5162_rtp_packetization I added the line "packetization = 30" for one peer in my sip.conf and started asterisk with the "-I" switch for async RTP. That's all it takes

Forgive this (possibly) silly question, but my upstream provider requires a packetization of 20ms. Using asterisk-oh323, I can set the "number of frames per RTP packet". How does this equate to packetization in ms?

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

Dear All, As per this article http://www.voip-info.org/wiki/view/Asterisk+cmd+Smsasterisk support PSTN to SMS and from SMS support endpoints to PSTN. In my scenario we have SS7 based E'1 on which our SMS provider sending SMS on our DID numbers and my all DID's are registered on OPENSER. What I want to do I want to receive SMS from PSTN on E?1 and forward on Register user?s if Register

Hello all, I am running asterisk on VMs with standby heartbeat configuration, Heartbeat assigns a virtual IP 172.20.255.40 on machine afterwards asterisk is started. In the sip.conf, I have explicitly define bindaddr=172.20.255.40 but sometimes I see packets coming from physical IP 172.20.255.41 I have both tcp and udp transport enabled Here is the lsof -ni :5060 output asterisk 2878 asterisk

Hi all, I am running an -MNP- multinomial probit model package using R. It gives me the following objection instead of giving me the results: Erreur : impossible d'allouer un vecteur de taille 137.9 Mo (in english: cannot allocate a 137.9 Mb vector memory).  I have already increased the memory size upto 2047Mb. This problem has been discussed in 2008 (archives) but no profitable answers were

To my knowledge, Asterisk's packetization rate is hard coded at 30ms. If I wanted to, where in the code could I go to change it to 20ms. Is there anything bad that might happen if I change it (asterisk related)? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/6088397a/attachment.htm