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remote ipphone

Hi

I would like to configure a single IPPhone on Site NORFOLK to use TWICKENHAM SIP no

both sites connected via VPN and can do intersite extension calling without any problem.

please see the diagram (TWICKENHAM router SIP has a spare number, so i would like to use it on NORFOLK phone for outgoing and incoming)

I entered following related commands on NORFOLK router

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

sip

  no update-callerid

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

!

!

dial-peer cor custom

name sip2800

dial-peer cor list usesip2800

member sip2800

!

!

dial-peer voice 2800 voip

corlist outgoing usesip2800

destination-pattern .T

session protocol sipv2

session target ipv4:83.245.6.81  **this the sip provider ip/sip registrar server ip. This ip is used in other side of the router to route the cals via sip)

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

!

dial-peer voice 2801 voip

corlist incoming usesip2800

answer-address 222 **extension on the phone***

!

!

ephone-dn  44  dual-line

number 222 secondary 0208123456

label sipcall

description sipcall

name sipcall

corlist incoming usesip2800

corlist outgoing usesip2800

ephone 24

button 1:44

mac-address xxx.xxx.xxx

AM i missing anything. is it possible. is there any better ways to accomplish this task

Basically when ephone 24 attached to NORFOLK router placed a outbound call the call should route through the TWICKENHAM router by using its sip no.

Regards

shameer

2 Accepted Solutions

Accepted Solutions

Hi Shameer,

You will need to do the following to get this to work, based also on the diagram you sent to me and the information supplied.

Site "B":

dial-peer voice 2031 voip

description ***ISR2800-CALLING-UC500***

preference 1

destination-pattern XXX

session target ipv4:

dtmf-relay h245-alphanumeric

!

dial-peer voice 2010 voip

description Incoming Call

incoming called-number .

Site "A":

dial-peer voice 2010 voip

description Incoming Call

incoming called-number .%

!

dial-peer voice 2015 voip

description ***UC500-CALLING-ISR2800***

preference 1

destination-pattern XXX

session target ipv4:

dtmf-relay h245-alphanumeric

codec g711ulaw

Now this is working on the assumption you have CME installed on the ISR-2800, if not then the only other way is to do it via Manual programming of the phone at site "B"

To do it VIA manual programming you would ensure that there is a VPN tunnel between the two sites, you would need to make sure that each site can route to all the subnets (Vlans I.E VLAN-1 Data, VLAN-100 Voice and VLAN-90 if the UC has a CUE VLAN).

On Site "B" phone, you would program in the TFTP address manually in the phones configuration options, this way it can shoot down the VPN tunnel and speak to the TFTP-SOURCE address of the UC-500.

Once it is done this way, you only need the DN to be created on the UC-500 side and any voice translation rules/profiles setup to have that SIP trunk DID routed to this DN (Extension).

These are the only two options I know of and have done in the past that work, there might be others but I have not done them or am not aware of them.

Cheers,

David.

Cheers, David Trad. **When you rate a persons post, you are indicating a thank you or that it helped, but at the same time you are also helping to maintain the community spirit - You don't have to rate posts and you wont be looked down upon :) *

View solution in original post

Hi Shameer,

This is a typo on my part, the session target should be different on both sides, for instance Site "A" should be pointing to site "B" and site "B" should be pointing to site "A" this is the only way it will work.

Site "A" should have said pointing to "ISR" and site "B" should have said pointing to "UC" my excuse is that I was tired and my brain was mashed up from a bad day at work managing too many problems... Well that is my excuse and I'm sticking to it

Cheers,

David.

Cheers, David Trad. **When you rate a persons post, you are indicating a thank you or that it helped, but at the same time you are also helping to maintain the community spirit - You don't have to rate posts and you wont be looked down upon :) *

View solution in original post

6 Replies 6

Hi All

Can you please help me on this

thanks

Hi Shameer,

You will need to do the following to get this to work, based also on the diagram you sent to me and the information supplied.

Site "B":

dial-peer voice 2031 voip

description ***ISR2800-CALLING-UC500***

preference 1

destination-pattern XXX

session target ipv4:

dtmf-relay h245-alphanumeric

!

dial-peer voice 2010 voip

description Incoming Call

incoming called-number .

Site "A":

dial-peer voice 2010 voip

description Incoming Call

incoming called-number .%

!

dial-peer voice 2015 voip

description ***UC500-CALLING-ISR2800***

preference 1

destination-pattern XXX

session target ipv4:

dtmf-relay h245-alphanumeric

codec g711ulaw

Now this is working on the assumption you have CME installed on the ISR-2800, if not then the only other way is to do it via Manual programming of the phone at site "B"

To do it VIA manual programming you would ensure that there is a VPN tunnel between the two sites, you would need to make sure that each site can route to all the subnets (Vlans I.E VLAN-1 Data, VLAN-100 Voice and VLAN-90 if the UC has a CUE VLAN).

On Site "B" phone, you would program in the TFTP address manually in the phones configuration options, this way it can shoot down the VPN tunnel and speak to the TFTP-SOURCE address of the UC-500.

Once it is done this way, you only need the DN to be created on the UC-500 side and any voice translation rules/profiles setup to have that SIP trunk DID routed to this DN (Extension).

These are the only two options I know of and have done in the past that work, there might be others but I have not done them or am not aware of them.

Cheers,

David.

Cheers, David Trad. **When you rate a persons post, you are indicating a thank you or that it helped, but at the same time you are also helping to maintain the community spirit - You don't have to rate posts and you wont be looked down upon :) *

Hi David

Thank you very much for the solution.

I  did implement the solution today and able to dial the extension and see its ringing.

But unfortunately there is no one at the other site to answer the call, also to do outbound calls.

so waiting for response from the customer. I'll update you as soon as i get some reply.

Hope it work

thanks

shameer

Hi David

just a small question

why we assign same ip address for session target on both SiteA and SiteB dial-peers

session target ipv4:

THANKS FOR YOUR HELP

SHAMEER

Hi Shameer,

This is a typo on my part, the session target should be different on both sides, for instance Site "A" should be pointing to site "B" and site "B" should be pointing to site "A" this is the only way it will work.

Site "A" should have said pointing to "ISR" and site "B" should have said pointing to "UC" my excuse is that I was tired and my brain was mashed up from a bad day at work managing too many problems... Well that is my excuse and I'm sticking to it

Cheers,

David.

Cheers, David Trad. **When you rate a persons post, you are indicating a thank you or that it helped, but at the same time you are also helping to maintain the community spirit - You don't have to rate posts and you wont be looked down upon :) *

Thanks David

I really appreciate your Help

Thanks

shameer

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