similar to: DTMF Problems

I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both

Having big problems and for months. Our service provider (via:talk) says they are Asterisk friendly but they are not. Here are the specifics (please read the bottom of the msg too) System: Dell SM Business server 2GB RAM, Core II Processor (should be plenty) OS: open SUSE 11 Asterisk Version: 1.4.2 Asterisk GUI Version: 2.0 The system was completely set up using the Asterisk GUI with a

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

Now that I have a new card and my echo problems are 'mostly' solved, I have another major issue to deal with. After about an hour or so the card will stop detecting DTMF tones on incoming calls. dahdi_monitor shows the following: [root at asterisk wctdm24xxp]# dahdi_monitor 1 -v Visual Audio Levels. -------------------- Use chan_dahdi.conf file to adjust the gains if needed. ( # =

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

It gets hard to read my logs when every time someone makes a phone call it displays long pages of "Dropping voice frame". Anyone encounter this before? Asterisk is bridging two SIP lines together, so the technology should be the same. Maybe I'll try allowing only ULAW. ************************************** Asterisk Standard debug (level 3)

Hi, I am trying to setup a simple home voip service w/ * I have compiled and installed the svn source as a first step I am trying to configure SIP for inside my network. I have a handful of softphones and a few hardphones that I want to all be able to call each other I have configured users.conf with a single softphone(kphone) and have tried calling itself (ext 6000) and the demo from the

I wish to have my Grandstream GXP-2000 phones make a different distinctive ring for internal calls ( Internal ) or if the incoming call has no caller id 'NOCID'. The Grandstream phones calls allow 3 distinctive rings depending on the caller id. I have one set up and working for 'Internal' calls but unfortunately the same tone will ring if caller id is absent on a call. My

Hi There, We have a Asterisk 1.4 box with a X100P card connected to a analog line with Caller ID serrvices enabled on it. When an incoming call appears we get the following in the log: -- Starting simple switch on 'Zap/1-1' -- Detecting post-CID distinctive ring [Apr 15 10:38:07] NOTICE[7151]: chan_zap.c:6469 ss_thread: Got event 18 (Ring Begin)... [Apr 15 10:38:07] NOTICE[7151]:

Hi All, Anyone have any idea what could cause incoming calls on a SIP channel to no longer be able to use DTMF? DTMF on incoming calls on zaptel and on local SIP softphones and ATAs all work fine. Nothing gets registered in the CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

Hello mailing list, I have been porting one of my Asterisk boxes to 1.4 and I have encountered a nasty DTMF problem. What happens is someone might come in to my IVR and enter "12345" and what will actually come through could be along the lines of "12234445". Sometimes it works, sometimes it doesn't. I had this problem with 1.2 back in November but was able to solve it

I have a simple dialplan. Using the read cmd, I ask caller for his passcode. If the caller is calling from an plain old analog phone, his dtmf is not heard until the prompt stops playing. SIP phones work correctly. I've trird everything I found searching the net. I've tried all the dtmfmode. I'm using 1.4.26 Currently my vitelity sip account is setup: insecure=very canreinvite=no

Hello, I have a problem of DTMF duplication. I receive call from my provider with SIP protocol. These calls pass through an interactive voice menu, using the application Waitexten to enter a client code. The menu works fine, but sometimes I have DTMF duplication that prevent proper code entry. All DTMF come twice. my sip.conf ----------- [general] context=default allowguest=no

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

Try as I might, I can not get incoming calls from ViaTalk to match against my user entry. I have both peer and user entries, and incoming and outgoing calls work, but incoming calls do not move to my in-viatalk context (they stay in the default context.) Has anyone else managed to get this to work? My user entry looks like: [viatalk-in] username=1407965XXXX context=viatalk-in type=user

Hello all, I know it was on the list before but i have some questions about the Kirk IP600v3, the requested configuration files were send private i guess Does anybody have the correct SIP settings for handsets connected to the Kirk. IP600v3 I am particulair intrested in settings regarding: -Voice Mailbox -Call waiting -DTMF settings for e.g. parking an extension with asterisk functionality

Hello all. I have a dahdi card with 2 FXO and 1 FXS. I'm able to make calls without any problem. However, when I have an incoming call, I see the following message on the asterisk console: -- Starting simple switch on 'DAHDI/1-1' -- Executing [s at DID_trunk_1:1] ExecIf("DAHDI/1-1", "1?SetCallerPres(unavailable)") in new stack -- Auto fallthrough,

We are having an issue where DTMF is not being sent out right away and the tone duration is inconsistent. For a test we send a '5', then a second later we send a '9', and then five seconds later we send a '5'. If you look at the logs below you can see the first '5' is played right away, then the '9' comes in and gets queued, but it doesn't start

> I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in > routing calls to upstream carrier via SIP trunks out.? I spent a lot of time > in the lab testing 1.8 which included heavily testing DTMF with no issues > that came up.? It all just seemed to work fine.? But then again you can?t > reproduce every real work scenario in the lab. > > > > I?m