similar to: How configure asterisk server extension.conf.

Hi All, Need Help ! I am interested testing opus codec encoding decoding qaulity. for this have complied opus code codec from souce. After compiling i got opus_demo app. for Encoding i followed below steps: 1) Reference file used music_orig.wav (http://www.opus-codec.org/examples/samples/music_orig.wav) Number of samples : 4358219 (90.8 s) 2015-04-13 10:40:07 UTC Sampling

Hi Sakharam, I see 2 potential issues with what you are doing. 1. ./opus_demo -e voip 48000 2 16 music_orig.wav testcase30.opus in above command, "16" for bits/sec seems too low. I'm no audio expert, but just cant convince myself you can get any reasonable audio data with 16 bits/sec. FYI, I was able to encode and decode with 16 bits/sec, but when I played the decode file with

Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29]

I'm trying to configure sip2sip, which says: http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk "Asterisk, is currently unable to handle more that one result for a DNS SRV lookup, and the Asterisk configuration needed for getting it work with the SIP2SIP service is not trivial" It then gives a complex multi-section workaround in SIP. I remember reading there'd be

Hello ALL, Anybody performed ASTERISK Testing for RFC 3261 Compliance? If Yes, Please share Result. Best Regards,Sakharam Thorat. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131028/1dcffece/attachment.html>

Is Asterisk is fully Complaint to RFC 3261 ? I am facing issue with expire Header , Contact header expire parameter, Max forward header range.?For e.g RFC 3261 say Max-Forward Header range should be 0-255 Data from RFC-> 20.22 Max-Forwards The Max-Forwards header field must be used with any SIP method to limit the number of proxies or gateways that can forward the request to the

Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1&t=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching and late nights since, but have had to admit defeat. Rather than duplicate it all here, I've posted my logs and conf files on that thread, too. Problem is that while

Hello, I cannot call my server over the internet with SIP anymore. Even when I do a maximum logging on my firewall, I don't see packets coming from outside. I've tried it from an ekiga.net account and an sip2sip.info account. What could be wrong? I would expect incoming traffic on port 5060 UDP... The account is "paul at vandervlis.nl". This should connect trought DNS to the

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm

Hello, We have a sip trunk between our voip operator and our asterisk 1.6.2.9 We have no problem during voice communications. But we can not send any t38 fax via this gateway. We tried to trace the error made some tests.. There are 2 main tests we tried to do. As i learned their voip path is like .. we connect to session border controller..then it routes the call to a cisco media gateway if the

Dear all, I have enabled stun module and configured it in asterisk , but asterisk not using stun returned public ip address for any of the sip requests going out of my network. i have done settings as below res_stun_monitor.conf settings: [general] stunaddr = stun.ideasip.com stunrefresh = 30 stun show status Hostname Port Period Retries Status ExternAddr

Hi, Does anybody know how codebooks are generated in OggVorbis encoder? We are porting oggorbis encoder on embedded platform for which VQ codebook memory is hugeeee to imagine. How can we reduce that? Can we do VQ with less codebooks and if yes how? If any help available? Parul Embedded Engineer Einfochips Ltd

We are implementing oggvorbis decoder on embedded system. In that we want to have support of chaining and grouping. If somebody can throw some light on how it can be implemented, it would be gr8 help. Regards, Parul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis-dev/attachments/20060523/d90e36a0/attachment.htm

Hello to All Members of this Group, Can I get the Test Vectors in .wav format for different sampling rates in order to Test the Ogg Vorbis Encoder Code ? If possible Forward the Link where I can get effective testvectors to Test the Encoder. Please Reply As soon as possible . Thanks & Regards , Maulik Desai Embedded Engineer - Embedded Division eInfochips Ltd. Work:

For testing the encoder i needed test vectors. thanks for the links send by members. Those links are useful, but contain test vectors corresponding to 44 KHz only. Does anybody has any idea where i can find test vectors of other sampling rates i.e. 48 KHz, 32 KHz, 16 KHz, 11 Khz, 8 KHz. ? Thanks, Parul Embedded engineer Einfochips

We are working on OggVorbis encoder. In the porting effort we are trying to convert it to fixed point code (both 32 and 24 bit fixed point). Now the issue is how we do the testing. What should be the criteria for our testing. Does anybody has any idea how compliance testing (i.e. some objective tests) is done at encoder side? What is the criteria of testing at the encoder side? If anybody

Probably doesn't help diagnose the problem.... but there were also audio problems experienced with this cvs version even on LAN / sip2sip / no transcoding > > ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... > > I will be looking into this issue later today. __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take

Hi Group, Is there any information available for Asterisk 1.8 dimensioning? I googled but couldn't find helpful data for 1.8. I am trying to figure out hardware configuration for following features implemented in Asterisk 1.8? (1)100 SIP clients. (2)ACD (Around 15 realtime queues) (3)Call recording for all SIP clients. (4)4 port PRI (E1). There would be around 100 concurrent calls.

On 5 February 2017 at 05:36, Jelle de Jong <jelledejong at powercraft.nl> wrote: > What do you guys tinc of wireguards, are there advantages? Jason seems to > have a good grip of what he is talking about. Well if it's kernel only, that rules out anything not Linux, at lest at the moment. I know that may have a big share, but I find that limit. I understand it being in the kernel

Dear all I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 but i got Noice on voice call so what would be the resone and how to fine tune my voice quality on asterisk ?? what codec would be best for my asterisk --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next