07-04-2015 03:02 AM - edited 03-17-2019 03:32 AM
If every call whether SIP or H323 requires 4 call legs, (2 from first router) (or ISP in some cases) and (2 from own router), how does an individual set up the dial peers for the ISP, unless they do this themselves?
07-04-2015 05:44 AM
Hi John,
Can you please elaborate it bit more what do you want to achieve...
Thanks
Vivek
07-04-2015 05:57 AM
Evening Vivek. Thanks for the quick reply.
I want to achieve receiving incoming calls through ip phone via DID number and outgoing calls via dial peers
Created incoming and outgoing dial peers for this. Upon testing DID number with calling via mobile phone. Call goes straight to voice mail. Phone not even ring.
Have verified with provider that am getting a 10 digit DID number
07-04-2015 06:04 AM
Hi John,
I think you have this discussion going on in other thread and had helpful comments. Didn't resolve yet?
https://supportforums.cisco.com/discussion/12537886/testing-incoming-dial-peer
Thanks
Vivek
07-04-2015 06:30 AM
Unfortunately no. I hadn't had any reply for 4 days now. I was happy with Jaime Valencia comments so done more reading. Hence the reason why I done the new discussion. Some times Cisco gives more unanswered questions than answers
07-04-2015 07:42 AM
I don't understand how I'm supposed to test an incoming dial peer from ISP to my router via debug voice dial-peer inout
I ran the command on router but nothing occurs
07-04-2015 08:05 AM
This show command would let you know which dial peers are being matched, please post "debug ccsip messages" for the call in question, so we can let you know if it makes it properly.
07-04-2015 08:11 AM
Hi Chris,
Thanks for your reply. Ran command but nothing happened.
07-04-2015 08:14 AM
Did you have "term mon" one and is term logging enabled?
Do you see anything for an outbound call as that is working per your description?
Posting "sh run" would be helpful.
07-04-2015 08:32 AM
07-04-2015 08:38 AM
As requested..
term mon output
Jul 4 15:34:20.791: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:sip1.exetel.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK2C822BD
From: "0300" <sip:0300@sip1.exetel.com.au>;tag=2F9C410-16E
To: "0300" <sip:0300@sip1.exetel.com.au>
Date: Sat, 04 Jul 2015 15:34:20 GMT
Call-ID: 36C05FFD-2BDE11D6-8002B05C-F4FCAA50
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 10
Timestamp: 1436024060
CSeq: 357 REGISTER
Contact: <sip:0300@192.168.3.1:5060>
Expires: 3600
Content-Length: 0
Jul 4 15:34:24.791: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:sip1.exetel.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK2C822BD
From: "0300" <sip:0300@sip1.exetel.com.au>;tag=2F9C410-16E
To: "0300" <sip:0300@sip1.exetel.com.au>
Date: Sat, 04 Jul 2015 15:34:24 GMT
Call-ID: 36C05FFD-2BDE11D6-8002B05C-F4FCAA50
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 10
Timestamp: 1436024064
CSeq: 357 REGISTER
Contact: <sip:0300@192.168.3.1:5060>
Expires: 3600
Content-Length: 0
07-04-2015 08:51 AM
Sorry Chris,
Im off to bed. Please leave any comments. :) :)
My aim is to find out why incoming calls aren't arriving. Antivirus Firewall is disabled. Have enabled ports on default gateway 192.168.0.1.
07-05-2015 05:04 AM
H Chris,
Sorry. Forget to include the translation rule for this
cme_router(config)#voice translation-rule 1 cme_router(cfg-translation-rule)#rule 1 /^0280070300$/ /0300/ cme_router(cfg-translation-rule)#exit cme_router(config)#voice translation-profile Outgoing-CME-To-Phonecme_router(cfg-translation-profile)#translate called 1 cme_router(cfg-translation-profile)#
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