«

Yes there is definately some equipment missing between the Asterisk and the
DMS 100 Switch. First of all a Nortel DMS has no VoIP Boards, it's all
PSTN. That's why I put a Cisco AS5300 connected to the DMS on the PSTN side
with 4 T1's and as for the VoIP side I need for it to register somewhere
where another device can route to it. So how can I get it to register to a
SIP proxy just like the Cisco ATA registers to the SIP Proxy?

Matteo D'Amato
Intelco Communication


Andrew Zhilenko <> wrote in
news:87GdnS801bbWfweiRVn-:

> Hello!
>
> shido wrote:
>
>> Cicsco ATA 186
>> |
>> Asterisk Box / Runs Linux / SIP Proxy / and can run your nat /
>> firewall ,
>> |
>> Telco Switch ( Nortel DMS 100)
>
> So, basically in answer to his question "how to route the call to
> AS5300?" you are proposing to him to install Asteriks, get rid of his
> AS5300 and connect T1s directly to the Asterisk. I won't argue about
> this solution, let the Matteo decide. But tell me - does it mean that
> Asteriks is not capable of routing SIP call from the ATA to remote
> AS53000?
>
> WBR, Andrew

Hello!

Matteo D'Amato wrote:

> Yes there is definately some equipment missing between the Asterisk and the
> DMS 100 Switch. First of all a Nortel DMS has no VoIP Boards, it's all
> PSTN. That's why I put a Cisco AS5300 connected to the DMS on the PSTN side
> with 4 T1's and as for the VoIP side I need for it to register somewhere
> where another device can route to it. So how can I get it to register to a
> SIP proxy just like the Cisco ATA registers to the SIP Proxy?
As I already said in the other message, you cannot. You can, however,
configure your SIP server that it will know, that some phone numbers (or
whole phone prefixes) should be routed to your remote Cisco GW. How it is
configured on the SIP server - depends on what kind of SIP server you are
using. My colleague says that in Asterisk it is possible to do it in the
config files. In the vovida SIP express router routing is done in a config
file, in a sort of mini-program. In the commercial products you normally
will get some interface (GUI or web) to conveniently manage your SIP routing.

WBR, Andrew
--
Andrew Zhilenko
Please remove "hide-email." from my email address when replying,
so my address should be andrew (at) ti dot cz

I found this link :

http://www.okena.com/en/US/products/...ucts_feature_g
uide09186a0080181111.html#1170970

and with the command

registrar {{dns: address | ipv4: destination-address} expires seconds
[tcp] [secondary]}


I think I will be able to register it, but one problem

Supported Platforms
Cisco 1751, Cisco 1751V, Cisco 1760, Cisco 2610XM, Cisco 2611XM, Cisco
2620XM, Cisco 2621XM, Cisco 2650XM, Cisco 2651XM, Cisco 2691, Cisco 3640,
Cisco 3640A, Cisco 3660; Cisco 3725, Cisco 3745


Why the heck didn't Cisco make this command available on a AS5300??

--

Matteo D'Amato
Intelco Communication


Andrew Zhilenko <> wrote in
news:lq-dnVuOt8zutgaiRVn-:

> Hello!
>
> Matteo D'Amato wrote:
>
>> Yes there is definately some equipment missing between the Asterisk
>> and the DMS 100 Switch. First of all a Nortel DMS has no VoIP Boards,
>> it's all PSTN. That's why I put a Cisco AS5300 connected to the DMS
>> on the PSTN side with 4 T1's and as for the VoIP side I need for it
>> to register somewhere where another device can route to it. So how
>> can I get it to register to a SIP proxy just like the Cisco ATA
>> registers to the SIP Proxy?
> As I already said in the other message, you cannot. You can, however,
> configure your SIP server that it will know, that some phone numbers
> (or whole phone prefixes) should be routed to your remote Cisco GW.
> How it is configured on the SIP server - depends on what kind of SIP
> server you are using. My colleague says that in Asterisk it is
> possible to do it in the config files. In the vovida SIP express
> router routing is done in a config file, in a sort of mini-program. In
> the commercial products you normally will get some interface (GUI or
> web) to conveniently manage your SIP routing.
>
> WBR, Andrew

Hello!

Matteo D'Amato wrote:

> and with the command
>
> registrar {{dns: address | ipv4: destination-address} expires seconds
> [tcp] [secondary]}
>
>
> I think I will be able to register it, but one problem
Please read carefully what it says in this document, just few lines above
the description of the registrar command:
....
To enable Session Initiation Protocol (SIP) gateways to register E.164
numbers on behalf of analog telephone voice ports (FXS), IP phone virtual
voice ports (EFXS), and SCCP phones with an external SIP proxy or SIP
registrar, use the registrar command in SIP user-agent configuration mode.
To disable registration of E.164 numbers, use the no form of this command.
....

> Why the heck didn't Cisco make this command available on a AS5300??
Because there are no FXS ports on AS5300.

WBR, Andrew
--
Andrew Zhilenko
Please remove "hide-email." from my email address when replying,
so my address should be andrew (at) ti dot cz

Yes, you are right. Well it might be related to the SIP Proxy server
itself, I will give Asterisk a try and I'll let you know what happens.
Thanks Andrew and Shido for your help.

--

Matteo D'Amato
Intelco Communication


Andrew Zhilenko <> wrote in
news:edidnTmdddHfoAaiRVn-:

> Hello!
>
> Matteo D'Amato wrote:
>
>> and with the command
>>
>> registrar {{dns: address | ipv4: destination-address} expires seconds
>> [tcp] [secondary]}
>>
>>
>> I think I will be able to register it, but one problem
> Please read carefully what it says in this document, just few lines
> above the description of the registrar command:
> ...
> To enable Session Initiation Protocol (SIP) gateways to register
> E.164
> numbers on behalf of analog telephone voice ports (FXS), IP phone
> virtual voice ports (EFXS), and SCCP phones with an external SIP proxy
> or SIP registrar, use the registrar command in SIP user-agent
> configuration mode. To disable registration of E.164 numbers, use the
> no form of this command. ...
>
>> Why the heck didn't Cisco make this command available on a AS5300??
> Because there are no FXS ports on AS5300.
>
> WBR, Andrew

Asterisk is very capable of routing a remote sip call to a remote 5300. It
can even transcode the codec and change protocols if you want.

.....Play an ivr before going out to the 5300
.......Record the call going out
...........Bill for the call

--
Greg Merriweather
The NuFone Network

519-251-8225 x 3000
IM:
"Andrew Zhilenko" <> wrote in message
news:87GdnS801bbWfweiRVn-...
> Hello!
>
> shido wrote:
>
> > Cicsco ATA 186
> > |
> > Asterisk Box / Runs Linux / SIP Proxy / and can run your nat / firewall
,
> > |
> > Telco Switch ( Nortel DMS 100)
>
> So, basically in answer to his question "how to route the call to AS5300?"
> you are proposing to him to install Asteriks, get rid of his AS5300 and
> connect T1s directly to the Asterisk. I won't argue about this solution,
> let the Matteo decide. But tell me - does it mean that Asteriks is not
> capable of routing SIP call from the ATA to remote AS53000?
>
> WBR, Andrew
> --
> Andrew Zhilenko
> Please remove "hide-email." from my email address when replying,
> so my address should be andrew (at) ti dot cz
>

Asterisk has T1 interfaces what equipment is missing? You can either plug
the smartjack directly into the interfaces or plug in a crossover t1 cable
and plug in a channel bank if all you have are analog lines... take your
pick.


--
Greg Merriweather
The NuFone Network

519-251-8225 x 3000
IM:
"Matteo D'Amato" <> wrote in message
news:Xns941FCB6003E58matdamatohotmailcom@216.40.28 .71...
> Yes there is definately some equipment missing between the Asterisk and
the
> DMS 100 Switch. First of all a Nortel DMS has no VoIP Boards, it's all
> PSTN. That's why I put a Cisco AS5300 connected to the DMS on the PSTN
side
> with 4 T1's and as for the VoIP side I need for it to register somewhere
> where another device can route to it. So how can I get it to register to a
> SIP proxy just like the Cisco ATA registers to the SIP Proxy?
>
> Matteo D'Amato
> Intelco Communication
>
>
> Andrew Zhilenko <> wrote in
> news:87GdnS801bbWfweiRVn-:
>
> > Hello!
> >
> > shido wrote:
> >
> >> Cicsco ATA 186
> >> |
> >> Asterisk Box / Runs Linux / SIP Proxy / and can run your nat /
> >> firewall ,
> >> |
> >> Telco Switch ( Nortel DMS 100)
> >
> > So, basically in answer to his question "how to route the call to
> > AS5300?" you are proposing to him to install Asteriks, get rid of his
> > AS5300 and connect T1s directly to the Asterisk. I won't argue about
> > this solution, let the Matteo decide. But tell me - does it mean that
> > Asteriks is not capable of routing SIP call from the ATA to remote
> > AS53000?
> >
> > WBR, Andrew
>

Hello!

shido wrote:

> Asterisk has T1 interfaces what equipment is missing? You can either plug
Great. The guy asks "how to route calls to AS5300?" so I suppose that he
already has it. In your "solution" Cisco AS5300 somehow is missing. Oh,
yes, who cares about that gray box about 10K worth, just stuck it under
the table and use it as a leg rest.

> the smartjack directly into the interfaces or plug in a crossover t1 cable
> and plug in a channel bank if all you have are analog lines... take your
> pick.
What if his SIP server (Asterisk or whatever) is in Montreal, and his PSTN
connection is somewhere in Albania? Take a veeeeeery long cable across the
ocean?

WBR, Andrew
--
Andrew Zhilenko
Please remove "hide-email." from my email address when replying,
so my address should be andrew (at) ti dot cz

Why would you do that? This is VoIP he would use the net.

--
Greg Merriweather
The NuFone Network

519-251-8225 x 3000
IM:
"Andrew Zhilenko" <> wrote in message
news:kKednXpOFYQlhgGiRVn-...
> Hello!
>
> shido wrote:
>
> > Asterisk has T1 interfaces what equipment is missing? You can either
plug
> Great. The guy asks "how to route calls to AS5300?" so I suppose that he
> already has it. In your "solution" Cisco AS5300 somehow is missing. Oh,
> yes, who cares about that gray box about 10K worth, just stuck it under
> the table and use it as a leg rest.
>
> > the smartjack directly into the interfaces or plug in a crossover t1
cable
> > and plug in a channel bank if all you have are analog lines... take your
> > pick.
> What if his SIP server (Asterisk or whatever) is in Montreal, and his PSTN
> connection is somewhere in Albania? Take a veeeeeery long cable across the
> ocean?
>
> WBR, Andrew
> --
> Andrew Zhilenko
> Please remove "hide-email." from my email address when replying,
> so my address should be andrew (at) ti dot cz
>