search for: audiomode

...hoppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don?t have this kind of problem on my Cisco 7960 which has VAD deactivated. The problem I don't see any VAD option in AudioModes of ATA. -- Juanjo sin .sig

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.

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

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!

...I am now trying to make it download the config file from the tftp server. I have set all of the options in the file and the file is definately named correctly. But the phone is simply not processing the config file for some reason. Two commands Im trying to get it o process is UIPassword and AudioMode It ignores everything though. I can post it here if you'd like

I have noticed a problem with the Cisco 7940/7960 phones where if you put your voice-mail box on hold using soft keys and come back you can no longer navigate. I am curious if anyone else can duplicate this problem. Happens reliably for me with the 7940 phones. I use rfc2833 for DTMF. I would think it was a Cisco bug, but for the fact that this did not happen with older version of

When I make a call using sip, I get the line NOTICE[327696]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec 19 received Repeated many times on the console ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to ;bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw

...'en') -- Playing 'vm-password' (language 'en') -- Incorrect password '' for user '123' (context = <any>) -- Playing 'vm-incorrect' (language 'en') -- Playing 'vm-password' (language 'en') - Tried to change AudioMode of ATA, to use 0x00140014, 0x11241124, and default 0x00150015. But there were no difference. - Tried to skip the password on *, using exten=>9999,1,VoiceMailMain(s123). * prompted there are new messages, and instructions on how to retreive them. Unfortunately, none of the dtmf keys worked (1...

...seems the ATA-186 wants to control the flash by putting my first call on hold and prompts me to dial another extension. DTMF is fine, just can't use the native Flash functions of our PBX with the ATA-186 and asterisk. Anyone done this or know which direction to go? I've tried changing the audiomode options, but nothing helped or I didn't get the right hex-to-binary conversion right. Any help would be greatly appreciated! Thanks!!

...: 0 StaticIP: 192.168.1.27 StaticRoute: 192.168.1.1 StaticNetMask: 255.255.255.0 EPID0orSID0: . EPID1orSID1: . CA0orCM0: 192.168.1.59:2727 CA1orCM1: 0 CA0UID: 0 CA1UID: 0 MGCPVer: NCS1.0 RetxIntvl: 500 RetxLim: 10 MGCPPort: 2427 CodecName: PCMU,PCMA,G723,G729 LBRCodec: 3 PrfCodec: 1 AudioMode: 0x00350035 ConnectMode: 0x90000400 CallerIdMethod: 0xc0019e60 DNS1IP: 0.0.0.0 DNS2IP: 0.0.0.0 Domain: . NumTxFrames: 2 TOS: 0x000068b8 OpFlags: 0x00000002 VLANSetting: 0x0000002b Polarity: 0x00000000 FXSInputLevel: 0 FXSOutputLevel: -4 SigTimer: 0x00000064 RingCadence: 2,4,25 DialTo...

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

Dear all, i have a problem with a system running - Ubuntu 10.04 ( all updates done ) - ii asterisk 1:1.8.5.0-1digium1~lucid Open Source Private Branch Exchange (PBX) - ii asterisk-dahdi 1:1.8.5.0-1digium1~lucid DAHDI devices support for the Asterisk PBX I also use freepbx 2.9 for the configuration. Hardware is a Dell R410 and a Digium

Hi, I'm using a Cisco ATA186 and iConnect to complete PSTN calls to the US. I've noticed that when I set the Cisco ATA to use LBRCodec to 0 (g.723 instead of g.729), AudioMode 0x00150015 and RxCodec, LxCodec to 0, (use g.723) Asterisk will connect to iConnect, successfully natively bridge the call and then about two seconds later not just drop the call, but terminate unexpectedly. The asterisk daemon will stop uncleanly. If I set the RxCodec, LxCodec back to 2 (g.71...

...y has to be configured to have it working fine. > Or this just happened to me? What is your ATA's software? > > 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833. None worked. > As per ATA, it is by default using rfc2833. I tried setting it up as inband > by setting Audiomode, but nothing helped. I was thinking the * is ONLY > recognizing the DTMF if there is telco board installed. Is it? > > > ----- Original Message ----- > From: "Philipp von Klitzing" <klitzing@pool.informatik.rwth-aachen.de> > To: "Jess Magnaye" <jess...

Hello, I'm looking for any comments or user experiences from anyone who is using 7912G phones with SIP. Any installation issues? Usability problems? Do the features seem to work, etc...In short, I'm looking for your opinions on how suitable this phone is for an asterisk implementation for approx. 10 users. Next logical question: what other phones would you recommend for a situation

Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound & get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered

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]: