[asterisk-users] No voice when the calls come from Internet
Hi, I have trouble establishing a call between between two SIP phones. One sip phone is, with asterisk server, at home behind a firewall. The second sip phone is an iPhone with 3G wireless connection. When I call from the SIP device at home the SIP account on the Internet (iphone + 3G) I can hear voice on both devices. But when I call from the Internet to my home SIP I get the ring but when I answer I don't hear any thing! I have asterisk v11 installed at home all the incoming TCP/UDP 5060 and a defined UDP range in rtp.conf forwarded to my Asterisk server. Do you have any idea when the voice is heard only when the call is from my local network to the Internet and not in the other direction ? Nevertheless, when both SIP devices are in the same home IP network the call is made without any problem whatever who starts the call. Regards, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
Hi Jeff, On 04/08/2014 12:13 PM, Jeff Brower wrote: Darrel- The G729 essential patents were *granted* in 1996, but applied for prior to June 8 1995. That means their lifespan is either 20 years from their application date, or 17 years from their grant date, whichever is greater (http://www.uspto.gov/main/faq/p120013.htm). Either way, they expire in 2014. -Jeff Where did you get the cutoff date of June 8 1995, and how does 20 years from that date lead to the last of the patents expiring in 2014? Nobody uses G.729. They use G.729A. The G.729A spec is somewhat later than the original G.729, but I don't know if there are any additional patents which specifically relate to Annex A. You could use G.729 instead, but it roughly doubles the compute needed. There are various things on the web saying the last of the patents on G.723.1, which was around in draft form long before G.729, expires in 2014. However, there seem to be patents related to that codec which don't really expire until some time in 2015. Its really hard to find solid information. The ITU patent database rarely identifies the actual patents being claimed, so its damned hard to look them up. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] .spec files or .src.rpms for packages.digium.com
Is it possible to get .spec or .src.rpms for packages on packages.digium.com? I specifically need to rebuild kmod-dahdi-linux-fwload-vpmadt032 for the kernels available in CentOS 5.10. I see there’s source at downloads.digium.com/pub/telephony/firmware/releases, but .spec files are not included. signature.asc Description: Message signed with OpenPGP using GPGMail -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP in dialog OPTIONS method handling
Hi everyone, I am running asterisk with release 12.1.0.rc3 and PJSIP. I have a peer which sends OPTIONS method for session keep-alive, and the asterisk is not responding to it. That of course disconnects the call after a few minutes. Is there a settings in the PJSIP.conf to respond to in dialog OPTIONS method? Looking at the documentation I haven't seen it. Does anybody know a workaround? Thanks, Yaron. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
