[asterisk-dev] SIP trunking

[Anton]

Today I've been trying to interconnect my network (*) with 
softswitch the NGN network of one of the local 3G telco ( 
Huawei SoftX ) via SIP. I've been requested "if my 
softswitch supports "SIP trunking" . I found a number of 
descriptions that such a thing like SIP Trunking is exist, 
but is it SIP-T (SIP, which also carry ISUP messages) or 
not exactly i did not find. I've been able to see a SoftX 
softswitch of the telco I've been trying to connect, but 
SIP behavior was very strange, I've heard "progress" - but 
when someone picks up the NGN phone on the telco side - My 
asterisk was starting playing me a MOH. So interconnect was 
unsuccessful. I found that latest open source project SipX 
3.6 support Sip Trunking (sipfoundry.org) and number of 
commercial SIP softswitches does too. So my today's 
experience means that there is already carrier SIP 
softswitches, asterisk is not interoperable yet.

Any ideas?

On 2 March 2007 13:13, Jonson Player wrote:
> What you mean SIP Trunking?
>
> On 3/1/07, Anton <anton.vazir at gmail.com> wrote:
> > Guys,
> >
> > any plans to support SIP Trunking?
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