[Asterisk-Users] SIP Testing

[Brian Capouch]

Stephen Davies wrote:>
> On 7 Apr 2003, Gregg Lebovitz wrote:
> 
> 
>>I would have someone explicity test against iconnect (I volunteer) since
>>I use this service quite a bit. I would like to see it finally work
>>properly. I have seen the problems with hangups reported on the list.
>>
>>Iconnect uses a Cisco Gateway.
> 
> 
> Please let me know how it goes.  I suspect it will now work right.
> 
> If it doesn't, please grab a "sip debug" of an example problem call and
> send it to me.
> 

I've made a dozen+ calls by now, all switched through perfectly.  I have 
the "|r" thing turned off; sometimes there is a short but noticeable 
delay even that way when the other party picks up.

I also cannot get DTMF through, with any of the three modes that I 
*think* are valid for that parameter.

I called one of my landlines to listen to the tones; they sound like 
little zipper-rips of white-ish noise.

I will take this over the 480 errors (which were always accompanied by 
phantom rings on the called party's instrument) any day.

Hopefully the rest will get ironed out.

Thanks.

B.