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Hi all, The problem is on the volume of the voice sent by the SPA-841. I think the echo cancel algorithm sets a limit to the microphone when detects sounds or noise from the earphone. This problem generates an oscillation on the voice volume sent by the phone and even turns it off completely for very little lapses of time making the communication very uncomfortable. I manage three different

Dear All, I am based in Australia and have a client looking to hire a VOIP Specialist with Asterisk experience to join their technical/engineering team. The company specialise in providing corporate and government grade data comms solutions. They are moving into the VOIP space, hence the need for the Asterisk/VOIP specialist. Would anyone be keen to explore this opportunity of working in

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

...e understandable diagrams of why SIP & NAT is so much bother. http://www.voip-info.org/wiki-STUN -------------------------------------- The VOIP INFO page about STUN - I don't think I learned much here - except the link to the Vovida STUN server software Asterisk Users - Email from wehr@japet.com - 02/July/2005 23:49 -------------------------------------------------------------------- Thierry claims that you need to put special MASQUERADE POSTROUTING rules into iptables to make it NAT UDP properly - tried it but didn't work for me Asterisk Users - Email from p_kami@yahoo.com - 16/...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

...>bother. >> >>http://www.voip-info.org/wiki-STUN >>-------------------------------------- >>The VOIP INFO page about STUN - I don't think I learned much here - >>except the link to the Vovida STUN server software >> >>Asterisk Users - Email from wehr@japet.com - 02/July/2005 23:49 >>-------------------------------------------------------------------- >>Thierry claims that you need to put special MASQUERADE POSTROUTING >>rules >> >>into iptables to make it NAT UDP properly - tried it but didn't work >>for me &...

Good afteroon i had found a special issue while using "sip show peers" sometimes i get 6115/6115 xx.xx.xx.xx 6109/6109 xx.xx.xx.xx 6001/6001 xx.xx.xx.xx 6107/6107 xx.xx.xx.xx 00/500 xx.xx.xx.xx 7000/7000 xx.xx.xx.xx in fact it must be 6115/6115

Good evening from the CVS of the 2005/05/14 it's impossible to build asterisk* on a redhat 7.3 i get this at compile time chan_sip.c: In function `build_user': chan_sip.c:10007: parse error before `struct' chan_sip.c:10029: `userflags' undeclared (first use in this function) chan_sip.c:10029: (Each undeclared identifier is reported only once chan_sip.c:10029: for each