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...src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not showing speex translation on "core show translation recalc 10". Can anybody please tell if missing some step in this. --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100806/eacc651e/attachment.htm

Hi, We have free pbx installed on asterisk 1.4.25.1. Mysql is installed and asterisk is connecting to it. CDR modules are all loaded as well. For some reason, it is not creating master.csv and no cdrs are generated. Can anyone help please. --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151 Email: deepika.nijhawan at oxygen8.com Skype: deepika-nijhawan W: <http://www.oxygen8.com/> www.oxygen8.com This communication contains information which is confidential...

Hi, Does anyone has an idea how to tell asterisk to use codec A for first 50 calls and then codec B for rest of the calls. Thanks, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/6114bf1d/attachment.htm

Hi, On some SIP interconnects with devices like Cisco, Dialogic we get SIP invite from different source port every time and asterisk rejects that INVITE. Does anyone knows solution for this? --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151 Email: deepika.nijhawan at oxygen8.com Skype: deepika-nijhawan W: <http://www.oxygen8.com/> www.oxygen8.com This communication contains information which is confidential...

Hi, I am trying to configure ipsec on asterisk. Have configured /etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in same folder. Have run racoon. Still I can't receive calls. Can anyone please tell if any extra step is needed. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am doing failover routing based on 2 dial commands. First route sends back 4xx response and I don't want it to try 2nd route when it is 4xx response. Can we do failover routing based on SIP 5xx response only ? Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

...users] Failover Routing Try this - it says it is for 1.8 but might work in 1.6 = http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Deepika = Nijhawan Sent: Tuesday, March 01, 2011 10:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Failover Routing SIP_HEADER() gives you only access to headers of the initial INVITE = request (and not, for example, the final BYE message) How will I check =...

Hi, I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. Dahdi status is not showing alarms but channels are not coming up. It is not showing any channels when i run 'dahdi show channels'. Could anyone help pls. Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, On freepbx (GUI), whatever reason number fails we always get 'all circuits are busy' audio. Does anybody know how to get far end audio when we dial wrong number or when it's busy or unallocated number or failed with some other reason. Thanks, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100615/e3f6ce8e/attachment.htm

Hi, Asterisk is making a call to a peer. In 200 ok, peer is sending its application server ip in contact field, so asterisk sends ACK to that IP. RTP starts flowing between endpoints and peer plays an IVR and asks for destination number. After entering destination number peer's application server sends INVITE again with different media IP and asterisk accepts with 200 ok. RTP from peer

TC400 and TCE400B digium cards that do codec translation, 1 How many of these cards can be installed on one server? 2 Can we combine it with the software codec aka for example if we had two cards per server they could decompress 240 channels. However if we had another say 100 calls on top of that (340 total), could the additional calls be passed over to the cpu to be converted? Can anyone