in asterisk 188.8.131.52 I get spurious DTMF recognition over SIP from an Audiocodes. I think the DTMF recognition is the Audiocdes fault, the Audiocodes log seems to say so as well, but I want to make sure, and fixing the Audiocodes is not an option..
> On Fri, 3 Feb 2012, Yaroslav Panych wrote: > >> What exactly commands I should invoke in AGI instead ofand >>? STREAM FILE returns only after file ends, this is not what I >> want. On Fri, 3 Feb 2012, Steve Edwards wrote: > Execute stream file i..
On Wed, Feb 01, 2012 at 06:47:49PM -0500, James Sharp wrote: > On 02/01/2012 02:17 PM, bilal ghayyad wrote: > >All; > > > >I heard from some friends that there are a dsl router that has Linux OS > >and it has asterisk on it, so the ip phone can regis..
On Thu, Feb 02, 2012 at 10:45:21AM +0200, Oguzhan Kayhan wrote: > all, > I was using dahdi 184.108.40.206.9 version for a long time. > We decided to upgrade to 220.127.116.11 a few days ago. > After that we started to have some problems with dahdi channels. > PS:DA..
On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour wrote: > all, > > Ive tried search this problem on the list… no luck… > > The case is: > > without externip/localnet config on sip.conf [general] my SIP trunk works, > but with no audio ..
On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munirwrote: > all, > > Im getting one way audio when calling over the SIP trunk i.e. end device > B (remote end of SIP trunk) can hear device A (softphone registered with > Asterisk) but device A cant hear dev..
On Fri, Feb 25, 2011 at 3:01 AM, Axellewrote: > yes xxx are numbers (not real letters x), its just obfuscation and > anyway its easier to recognize them by the first few digits. > and yes, they match the phone. > Show us.The error youre receiving specifica..
On Thursday 24 Feb 2011, Edwin Quijada wrote: > Hello! > I bought a virtual IP line to my ISP to use with my asterisk but when I try > to connect it to my ISP tells me I can not use and I can only use with a > softphone that gives me, xlite ready configur..
sorry i wasnt clear enough i meen inbound On Thu, Feb 24, 2011 at 12:25 PM, Rizwan Hisham wrote: > use the timeout option in the Dial application like so > > Dial(SIP/trunk,120) > > If you dont specify the timeout the default timeout used bya ster..