01-06-2011 07:35 AM - edited 03-16-2019 02:43 AM
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
01-06-2011 08:43 AM
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.
01-06-2011 09:45 AM
TKS Giordano!!
All my POTS are with "no sip-register", why the extensions still appear on line colum?
Tks
01-06-2011 10:18 AM
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
01-06-2011 10:44 AM
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
01-07-2011 04:29 AM
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 !!!
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