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Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100323/da2eea19/attachment.htm

Hello, Does asterisk do transcoding when the call goes through the system, codecs are the same but signaling protocol is changed. example: SIP with GSM ---> IAX with GSM What quality destruction happen when I use transcoding? I know this is not a concrete/precise question, but I would like to know how is it in general. What CPU performance is needed for transcoding 30 channels e.g. from

Hi, I have just been playing with the latest X-Lite.. It works fine with Asterisk.. As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only one that didn't work.. not sure why.. But the bigger problem is that when I call another extension that is using a Snom200 the call connects but there is no audio in either direction.. I have tried G.711a/u and GSM and while X-Lite shows that the correct cod...

Hi All, Can someone explain to me the difference between g711's ulaw and alaw codecs? Is it just different header info or is the actual payload in each encoded differently? I have thus far noe been able to find any difinative information onthe matter. All I've managed to find out that they are "similar", they sound the same and that it doesn't matter which you use. Could

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the

hi, i have connected a SNOM 200 to the asterisk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process):...

Hi I'm setting up a SIP gateway, serving quite a few potential users, and I wonder if I should purchase a Opteron or Xeon based system. Xeon has it's HT, but is it worth it? Has anyone tested Opteron on asterisk? Does it work well? There'll be no transcoding in this system - G.711A all way through roy

...186's but not sure if it was 100% or not. I want to know really is it possible to run GSM or ilbc on them or is a G729 lic the only way to get a low bandwidth codec? This is the list of codecs that I have seen. RxCodec and TxCodec?Configure the codec ID. * G.723.1?Codec ID 0 * G.711a?Codec ID 1 * G.711u?codec ID 2 * G.729a?codec ID 3 Thanks David

...all of my dialplan logic in Asterisk regardless of the ATA in use. Am I right in saying that because Asterisk has Answer()'d the call and done DISA(...), I can't do a re-invite to bridge the call between the PAP2 and the VoIP provider? And even if I could, I couldn't set it up to use G.711a between the PAP2 and Asterisk, and the switch to G.729 when the call bridges to the VoIP provider? This would be useful in that I can use G.711a locally where I have the bandwidth, and I wouldn't need to get licenses to use G.729a (because asterisk wouldn't need to touch it). Assuming I...

...I have is lagging. What codec should I use? I have Cisco phones 7960 on both end. Currently in the IAX trunk I configured it to disallow all and use GSM only. In my sip config of each phone I use disallow all and allow ulaw and alaw only. I see that the Cisco phone support G.729a, G.711u and G.711a codecs. I know that Digium is selling G.729 codec on there website. Should I get that to fix my problem? The less bandwitdh the codec takes over the Internet the better chance I have that it will work fine I presume. The maximum upload speed I can reach is 64k/sec so 512kbits. But that's...

Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. If I choose G.729A the "pass-throu" calls among users are OK, but Asterisk can't transcode GSM to G.729A in voicemail calls. My company doesn'y want to pay for a G.729

...ence for total bandwidth usage - for example for an 8k codec like g.729 (or speex quality=3) trunking reduces bandwidth usage nearly in half). I'm probably being a bit optimistic about my accuracy in quoting results with the decimal place! Here are some results for codecs other than Speex: G.711a: 5 concurrent calls: 67.8kbps/call 30 concurrent calls: 66.2kbps/call G.729: 5 concurrent calls: 13.1kbps/call 30 concurrent calls: 10.8kbps/call iLBC: 5 concurrent calls: 16.6kbps/call 30 concurrent calls: 15.2kbps/call Speex is very versatile, so I started testing Speex...

...ively homogenising the SDP from both sides even if it subsequently removes itself from the media path! So, I end up with situations where on the one side, I get, say: Customer MGW --> OpenSER --> Asterisk - sends call as G.729. Asterisk --> OpenSER --> Our MGW - our MGW prefers G.711a. Now, if customer MGW <-> Our MGW were talking directly, as they do when the deal is brokered through the OpenSER proxy, they would simply negotiate upon what they agree. But for some reason with Asterisk this does not seem to be working as advertised; we get lots of failed calls if we pas...

...<----->LAN<----->Asterisk<--------> PSTN I'm restricted to use passthru mode for faxing, instead of T.38 protocol, because the Asterisk box is running v1.2 and cannot be changed as it is in a heavy production environment. Anyway, it "should" work in passthru mode (G.711a) as the ATA and the Asterisk are in the same LAN with very low traffic. The problem arises when I try to send a fax: the Asterisk server initiates the call and, after a few seconds, the Linksys hangs the call by sending a BYE message: DEBUG[7416]: chan_sip.c:11375 handle_request: **** Received ACK...

...------------------------------------------------------------------------- node001:~# wget http://apt.puppetlabs.com/puppetlabs-release-lenny.deb --2012-10-12 02:12:12-- http://apt.puppetlabs.com/puppetlabs-release-lenny.deb Resolving apt.puppetlabs.com... 96.126.116.126, 2600:3c00::f03c:91ff:fe93:711a Connecting to apt.puppetlabs.com|96.126.116.126|:80... connected. HTTP request sent, awaiting response... 200 OK Length: 3396 (3.3K) [application/x-debian-package] Saving to: `puppetlabs-release-lenny.deb'' 100%[===============================================================================...

Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729

...phone, I get a mashine gun noise and the following msg in asterisk log: NOTICE[262161]: File channel.c, Line 1091 (ast_read): Dropping incompatible voice frame on H323:0 of format SLINR since our native format has changed to ULAW Both, the Planet phone and the asterisk oh323 channel, have G.711A as preferred codecs. For me, it seems, that the planet phone does not follow that hint, when receiving a call. But shouldn't asterisk be capable of understanding SLINR? Any help is appreciated! Roger.

Hi, is there a definite answer if asterisk can pass calls between SIP and h.323 protocols? Thanks, Yiannis.

I just installed asterisk@home to see how it works and seems there is a problem with sounds... I dont hear any announcements or recordings... sounds are on /var/lib/asterisk/sounds and the logs show this: -- Created MeetMe conference 1023 for conference '8200' -- Playing 'conf-onlyperson' (language 'en') But I dont hear anything... any experiences with this kind

...art/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 -> * -> g729/H323 PS intel's g729 was used. ast 1.0.3-6 PPS stupid - h323_set_capability(format/*=8*/, dtmfmode); + h323_set_capability(capability/*=8+256 (711a+729)*/, dtmfmode); lead to segv only.