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Calls out dont work

Thiago Cella
Level 1
Level 1

People,please help me,

My router is registred with sip trunk provider , but i canot make calls out , i saw with the command "sip-ua register status" that there are some peers,

why that?

And why the my ID of the sip account is the peer -1?

Line                              peer        expires(sec)  registered
================================  ==========  ============  ==========
200                               20002       15            no
201                               20001       16            no
202                               20003       16            no
203                               20004       16            no
204                               20005       16            no
205                               20006       16            no
206                               20007       16            no
207                               20019       16            no
208                               20009       17            no
209                               20008       16            no
210                               20010       17            no
211                               20011       17            no
212                               20012       17            no
213                               20013       17            no
214                               20014       17            no
215                               20015       17            no
216                               20016       17            no
217                               20017       17            no
218                               20018       17            no
219                               20021       17            no
220                               20034       35            no
551111114820                      -1          135           yes

Tks

5 Replies 5

Cisco gateway tries to register any POTS dial-peer.

Do you have these pots dp in your config?

To disable registration process use the command no sip-register on every pots dial-peers.

To debug your issue use "debug ccsip messages" during call and post the result with gateway config.

Regards.

TKS Giordano!!

All my POTS are with "no sip-register", why the extensions still appear on line colum?

Tks

Hi Thiago,

The dial-peers which you can see on the SIP registration are for the ephone-dn's.

You need to add "no-reg both" in ephone-dn part of your config which relates to number.

for example:

number 2000 no-reg both

Chris

Tks Chris,

Now the only that apper is the ID of account:

551111114820                      -1          135           yes

But when i try make a call, the busy signal returns. I talked with the voip provider, and they said that when the router send th register with server sip , the router send the header with ID and pass correctily, but when i try make a call, the router dont send the ID and pass, what could be? I tried reboot the router , but didnt work!

Tks

Hi People,

Now its working, the problem was with the codec, the dial-peer was that :

voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte

I changed to :

voice-class codec 1
session protocol sipv2
session target ipv4:200.143.XXX.XXX
dtmf-relay sip-notify

Tks !!!