Is anyone out there using Asterisk to talk SIP with Verisign SIP-7 (SIP -> SS7 gateway service)? I'm looking to control some Cisco AS5400 MGCP gateways but I need SS7 with Verizon. Signaling would travel this path: PSTN - ss7 -> Verisign - sip -> Asterisk Bearer traffic would travel this path: PSTN - imt -> Cisco AS5400 -> ip/voip -Matt -- Matthew S. Crocker Crocker Communications, Inc. Vice President PO BOX 710 Greenfield, MA 01302 P: 413-746-2760 F: 413-746-3704 W: http://www.crocker.com E: matthew@crocker.com
Matthew Crocker wrote:> Is anyone out there using Asterisk to talk SIP with Verisign SIP-7 (SIP > -> SS7 gateway service)? I'm looking to control some Cisco AS5400 MGCP > gateways but I need SS7 with Verizon.Not yet, I haven't been able to get anyone from Verisign to contact me, even though I know people who know people there... They are either very swamped with business, or extremely disorganized.> Signaling would travel this path: > > PSTN - ss7 -> Verisign - sip -> Asterisk > > Bearer traffic would travel this path: > > PSTN - imt -> Cisco AS5400 -> ip/voipExactly. Verisign also supports Veraz and Audiocodes gateways at least, if not others.
>> Can you expand ... I'm hoping to use Asterisk to do SIP to MGCP >> mediation ... what does it not work?> I don't know the particulars, because I've never used (or even looked at > MGCP). All I know is that whenever the issue comes up, people here say > that Asterisk does not know how to act as an MGCP Gatekeeper, only as an > agent. I presume it would have to act as a gatekeeper to control an > MGCP-based media gateway, because those devices are all intended to be > controlled by some sort of softswitch.IMO, there is no such thing as an "MGCP gatekeeper"; try that phrase with Google and it will be obvious. "Gatekeeper" is an H.323 term. MCGP is a master-slave protocol. The master is referred to as a "Call Agent", a "Media Gateway Controller", or just a "softswitch". This is the role that Asterisk can play. The slave is a "Media Gateway", an "MGCP phone", an "MGCP ATA", or just an "endpoint". Asterisk cannot presently act as a slave. Of course, any large system may have higher-level elements that handle authorization, accounting, complex routing, queueing, etc., but those topics are beyond the scope of MGCP. Perhaps the term "gatekeeper" was used in that context. So, I think that Asterisk will provide the functionality that you desire. However, I don't know if SIP<->MGCP calls can presently be completed without Asterisk proxying the media stream, so you may have performance issues. Perhaps someone else can address that. --Stewart
> In what context will Asterisk will require proxying the media stream?> I have a simple setup whereby I make my FWD account ring my Mediatrix 2102 > as an extension to my Asterisk and the delay is horrificIf FWD is speaking IAX and the Mediatrix is SIP, * must remain in the loop, even in theory. If both are SIP, reinvite is enabled, and you meet the conditions for reinvite to be applicable, media should bypass *. Even though H.323 and MGCP have functionality equivalent to reinvite, I believe that * does not support it. Of course, if both sides are IAX, the transfer occurs automatically. --Stewart