Re: [Asterisk-Users] SPA-3000 is translating vocal sounds into DTMF

2006-01-06 Thread Mike Bernson
I am having the same problem with a male voice at the other end. It is making the spa3k problem for me. Has this been reported to SIPURA ? Is this a common problem ? has anyone done been able to make this happen less often ?On 1/6/06, Brian Roy <[EMAIL PROTECTED]> wrote: On 1/6/06, Brian Capouch

Re: [Asterisk-Users] Sip man in the middle

2006-01-01 Thread Mike Bernson
I am planing on doing it a daemon that can live on the asterisk box or any box that can run unix and iptables. I will need to reroute packets aimed for providers box to the box where the daemon lives. In my case using a low power(15watts) is the way to go. If your asterisk box has the spare power t

[Asterisk-Users] Sip man in the middle

2005-12-30 Thread Mike Bernson
What I am looking to do is the follow: provider ---route/iptables---sip hardware from provider   |   |   asterisk box. What I would like to be able to do is have the provid

Re: [Asterisk-Users] SIP and echo cancel

2005-12-19 Thread Mike Bernson
In this connection there are min of 2 hybrid circuits on my end Vonage ATA box SIPURA ATA Asterisk - SIPURA ATA (phone) hybird 1   -  2 wire -- hybird 2 --hybird 3 (if not 841 phone) I also think vonage has one more hybird on there end. Since the connection

[Asterisk-Users] SIP and echo cancel

2005-12-17 Thread Mike Bernson
I known that sip channel should be free from echo. I am find this is not the case for me. The setup here is Sipura 3000 connected to vonage extensions are SIPURA 841 or SIPURA 2002 ATA. I am getting echos on some of the outbound calls. I would like to be able to have one of the software echo canc

[Asterisk-Users] SIP REGISTER

2005-11-12 Thread Mike Bernson
From the dump that I have attached It looks like the first attempt at register does not work then followd by a second register which then works. This is happening on all the SIP phone attach to asterisk. The version of asterisk here is 1.2.0b2. Here is sip.conf for ext 204 [204] username=204

[Asterisk-Users] Problem with Aterisk 1.2.0 beta 2 and sip dtmf

2005-11-06 Thread Mike Bernson
I have a number of sipura 2002 ATA connected to asterisk. I have set them up with 'dtmfmode=rfc2833' for handling dtmf. I then setup setup an extension to test sending dtmf tones. exten => *40,1,Answer exten => *40,n,Wait(2) exten => *40,n,SendDTMF(123456789,500) exten => *40,n,Hangup When a C

Re: [Asterisk-Users] Problem with asterisk 1.0.9 and sip and dtmf

2005-10-23 Thread Mike Bernson
Silence Supp Enable is No. Sergey Okhapkin wrote: Check if you have Silence Suppression disabled on PSTN line of spa-3000 (admin/advansed/PSTN line). On Sat, 2005-10-22 at 18:05 -0400, Mike Bernson wrote: I have asterisk running with sipura 3000 connect to PSTN and sipura

[Asterisk-Users] Problem with asterisk 1.0.9 and sip and dtmf

2005-10-22 Thread Mike Bernson
I have asterisk running with sipura 3000 connect to PSTN and sipura 2000 connected to phones. On inbound calls I am getting what sounds like DTMF tone when someone is talking on the PSTN side of the phone. It sound like someone is hitting key on the phone while talking. Is there any way to stop