similar to: RFC-3389 problem

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

Hello All, We are running Asterisk on a linux server as a SIP proxy with Cisco ATA 186's at the subscriber end. For long distance we have iax2 connectivity with a ip carrier. For local calls we are routing out through a commercial VEGA voicestream pots unit to an adtran channel bank and then from there to our class 5 soft switch. The sip to sip calls and the long distance calls work great.

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

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

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 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]:

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

Hi guys, I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk. Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller. Immediately, I though into an RTP-Proxy problem, but is not. Then I saw that message appear on the Asterisk CLI, during

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

> 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,

Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx -> the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr

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

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 =>

I have an Asterisk 1.4 box which is sometimes getting the message below. Here is the weird part, the CNG is coming from ANOTHER ASTERISK SERVER. 209.220.119.19 is an Asterisk 11 box. [Oct 8 11:59:27] NOTICE[20798]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 209.220.119.19

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com

I am trying to set a up mail server on a ubuntu edgy machine. I followed the howto at http://wanderingbarque.com/howtos/mailserver/mailserver.html . I am able to configure everything except dovecot-ldap authentication :(. After struggling with it for a week I decided to seek help. The current version of dovecot is v1.0 RC21 and the version used for the howto is v0.99.10.9. I installed &

Hi to all, I have configured in my * a avm fritz card and in my isdn connection I have only 1 extensions. I read from documentation tht is possible to use the exten,fax to redirect an incoming fax. Is it possible also with avm card or only with digium card. I used in this way but with no success: exten => fax,1,Goto(fax,2202,1) 'Configured for spandsp exten =>