Thanks for reply. What if i create a dialpeer:

R1

Dial-peer voice 222200 voip

detination-patern 222200

session target ipv4: R2-IP

R2

dial-peer voice 222200 pots

destination-patern 222200

foward-digit 2

port 1/0:15

what u think using this above configuration, is it not possible? I believe as user press 222200 routet R1 will match dial-peer and forward to R2. R2 will match dialpeer and send just 00 to PBX2. As PBX2 recieve 00 it will seize pstn trunk so because of just pressing 222200 a trunk will be establish from user and pbx2 and as user press remaining digits, that will simply pass to pstn over voip. I think not need to add "T" in destination-patern. If i am wrong please correct me.

best regards

Hi,

this will work if PBX2 has analog lines to PSTN.

It would cause two stage dialing with second dial-tone from pstn.

But, if PBX2 has ISDN to PSTN, it will not work, because PBX2 will send a call setup with no called number to PSTN, and this generally causes an error.

I still do not understand why you think that a regular setup with T will not work, it should,

Friend,

i did not say that "T" is not work. But with T users have to wait for interdigit timeout or press # that is not acceptible to them thats why i dont go for it. Ok tell me how we can tweak this timeout so that user dont feel and delay after dialing pstn number of any length.

Secondly, i tested the same type of configuration with pri and it worked. For example:

phone1---pbx1---router1------router2----pbx2---phone2

pbxs are connected with router with e1 pri and routers are connected with each other via e1 link. Site1 code is 1000 and site2 code is 2000

site1 config

dialpeer 2000 voip

destinationpatern 2000

session target router2 ip

site2 config

dialpeer 2000 pots

destination 2000

port 1/0:15

forward digit all

so user dial 2000 and extention of any length, it worked properly and no error as you said. I would tell you one thing, in real i am adding "signaling forward rawmsg" in all voip DPs and "supplementary-service pass-through" in pots DPs.

Waiting.....

Hi,

I would like to see the q831 trace for the config you mentioned. To see if it is sending the digits in band or with info message.

Anyway, as you know the interdigit delay, when set to 3, is not much of a problem.

The alternative is that you build a dialplan that matches all or most numbers length .

usually that is not very difficult especially if it is about one single city. Get a phonebook and you will be able to rebuild the numbering tree, even if you country has variable-length numbering, it can be done .

Dear p.bevilacqua ,

Thanks for reply. you can find debug isdn q931 output in attachment.

Routers Configurations:

Router A

dial-peer voice 99 pots

destination-pattern 99..

supplementary-service pass-through

direct-inward-dial

port 1/0:15

forward-digits all

dial-peer voice 8701 voip

destination-pattern 8701

session target ipv4:10.0.0.2

Router B

dial-peer voice 99 voip

destination-pattern 99..

signaling forward rawmsg

session target ipv4:10.0.0.1

dial-peer voice 8701 pots

destination-pattern 8701

direct-inward-dial

port 1/0:15

forward-digits 0

Note: In dialpeer 99 you see i define 99.. where .. is because i have an other dialpeer with destination-patern 9900 thats why so don't confuse it. See i just define 99.. and actual number dialed is 9935240 and call is made properly no issue at all.

Waiting for your immediate response.

Best Regards

Hi,

yes in the trace you sent the router is doing proper overlap dialing. If you manage to reproduce this when calling to PSTN, there should be no delay, once you have passed enough digits to PSTN you should receive connect and the call goes through.