SIP Congestion

[ranjithvkumar]

Now all my outgoing calls are working fine. I have problem with incoming. From service provider i am receiving 10 digits and i need to remove the 1st digit from that. I tried to do this in ARS=>ARS route list=> Nb.Digits to be removed = 1 but still it doesnt work. And i tried with Ext.Call Back translation rules=> Nb.Digits to be removed = 1 but still it didnt work.


Can some expert here can tell me where else i can remove the 1st digit out of 10 digits which my service provider is sending me.

Thanks.

[haroun]

try to change at trunkgp /trunkgp level
number of unsded digit isdn=6
mean from 0123456789 only 6789 will be used for did ,

[haroun]

ah you need toremove the first one
so number of unused digit isdn =1

[haroun]

for did go translator/exernal
ddi numbering translator : create new one with number 2 with name etc
ionce creted go down hierarchy ddi numbering translator rule and create a rule according to what service provider ask (you use 9 from ten digit as ddi)
DID num. transl. identifier : 2 ?
? First External Number : 165299200 ?
? ?
? First Internal Number : 100313 ?
? Range Size : 1 ?
? Unique Internal Number + NO


GO to translator /external/npd
NCONSULT MODIFIE npd id =10
escription identifier : 10 ?
? ?
? Name : IU_IU_E_E_DDI ?
? Calling Numbering plan ident. + NPI/TON: ISDN Unknown ?
? Called numbering plan ident. + NPI/TON: ISDN Unknown ?
? Authorize personal calling num use + True ?
? Install. number source + Entity source (not changing) ?
? Default number source + Entity source (not changing) ?
? Called DID identifier :2 ?
? Calling/Connected DID identifier : 2

at entity put the istallation number 0165299200

[ranjithvkumar]

Now incoming calls are ok. the problem was in connection cos. But incoming calls dont have voice. for outgoing voice are ok.

I think problem with codec.

Service provider - G711 A-law so in trunk i mentioned as G711 but the incoming call voice is not working.

Any idea?

[haroun]

TRY IN SIP EWTGW
only trunk group algo + False ---> true

[ranjithvkumar]

TRY IN SIP EWTGW
only trunk group algo + False ---> true

I changed as u said but still no voice in incoming.

[haroun]

and in the traces you see the g711 as unique offred codec in sdp
please past the sip trace for incoming call with no voice or a wireshark capture .

[haroun]

may be incoming rtp ports are blocked
i don't suspect codecs because if it was the case the call should be released an no rtp media should be proceeded 0

[ranjithvkumar]

Please find the traces for the incoming calls.

May i know the port number range for RTP ?