similar to: sip tone question

Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263

I'm getting some warnings: NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Asterisk Version: CVS-01/06/04-13:50:26 Cisco ATA 186 version: v3.0.0 atasip (Build 031210A) Is this something I should be concerned about? Anyone know how to "turn off" the RFC3389 support on the ata 186? Thanks!

sorry to bother with this trivial issue, but i am loosing all my hair ;-) got 2 x100p's and * on a slakware box x-lite to x-lite works fine! i also have: #ztcfg -vvv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. and in extensions.conf i got: [locals] exten

Hi, Below if the error message which I got from asterisk. I was trying to fax to asterisk from my fax machine. I really dunno what is the problem. I use alaw & ulaw codec only through my ATA 186. Can anyone help me what could be the problem. -- Executing Goto("SIP/-080ef9a0", "13732|s|1") in new stack -- Goto (13732,s,1) -- Executing

-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 -- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 RFC3389: 5 bytes, level 0... Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Killed Whenever I make a call between extension 101 and 1009 which are both Xten Xlite SIP clients, I get that error and

I can't login anymore... used to be able to. Timing doesn't seem to be working well any ideas? Also what is this "NOTICE" I'm getting? *CLI> == Accepting call on 'SIP/lenny-b19c' ("Lenny Tropiano" <5555>) -- Executing VoiceMailMain("SIP/lenny-b19c", "") in new stack == Parsing

Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

> Hello all, > Can someone help me on the problem which I have on MGCP phone test . I test mgcp - asterisk- zap. But I got several NOTICE message from rtp.c. > NOTICE[20501]: File rtp.c, Line 221 (process_rfc3389): RFC3389 support > incomplete. Turn off on client if possible > > -- Endpoint 'aaln/1@VG101-1-1' observed '9' > NOTICE[20501]: File rtp.c,

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

Hello all ! I have a problem with my IVR with terminate connection from PSTN phone Here is my configuration extension.conf [ivri] ;exten => s,1,Wait(1) exten => s,1,Answer ;exten => s,2,DigitTimeout(5) ;exten => s,3,ResponseTimeout(10) exten => ivr,1,Background(demo-congrats) exten => 1,1,MP3player,/mnt/linux/mp3/song/04.mp3 exten =>

When making or receiving a SIP call via my service provider, I get the following message logged by Asterisk: Dec 11 15:13:37 NOTICE[7392]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Since the "client" is at my service provider (who uses CISCO kit, I believe), I don't have the

hello to all, i am geting this NOTICE while i am running asterisk. agents are able to here the customer voice but the customer is unable to here agent voice plz somebody help me #rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.230 -- M. VIDYASAGAR -------------- next part -------------- An HTML

I'm having problems using the following. [sip] exten => _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM}) [macro-test] exten => s,1,Answer exten => s,3,Flash exten => s,3,Dial(SIP/${ARG2},30,t) exten => s,4,Dial(SIP/${ARG1},30,t) exten => s,t,Hangup exten => s,i,Hangup exten => s,h,Hangup I know I must be missing something simple, but here is the output from

Hi all. I have some problems to hear clearly music on hold, the sound interrupting. this some logs what i have in asterisk : rtp.c:298 process_rfc3389: RFC3389 support incomplete RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1

Hi there, i had installed on all my servers the codec_g729b which is the old voiceage, so a month ago i updated the codec to codec_g729a. After that i started to get this message on my asterisk console: Dec 16 09:19:26 NOTICE[1288699200]: rtp.c:293 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256 Dec 16 09:19:26 NOTICE[1288699200]: rtp.c:264 process_rfc3389: RFC3389

I am getting (with older image): RFC3389 support incomplete. Turn off on client if possible How do I turn that off for the DG104s? Or if I can't how do I tweak asterisk? I see posts about ATA-186's having an audiomode, but the closet I came to was inbanddtmf. I tried =0 and =1, no effect. Thanks! -- Zot O'Connor <zot@zotconsulting.com> White Knight Hackers, Inc.

New install, Calls are working phone to phone using gsm, ulaw or alaw codec but when try and echo test or voicemail there is no playback. I've tried turning on and off every codec and still no luck. Asterisk says it's playing the sound file but I just don't hear anything. I can't find any reason for this. I've tried the latest tar and CVS with the same result.

My configuration is: [ISDNPRI] -- [CISCO AccessServer AS5350] --<H.323>-- [ASTERISK] -- [CISCO ip phone 12SP+/Skinny] When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN everything working ok (RTP is ok). But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone IP phone party can hear ISDN party, but ISDN (incoming) party canNOT hear IP phone party

Try sending 5350 config and oh323.conf, versions, etc... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Goran Dj. Sent: Wednesday, December 22, 2004 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party