cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
1661
Views
0
Helpful
7
Replies

Sip trunk and Fios

rdossouvi
Level 1
Level 1

     Anyone has a luck setting up Sip trunk with FIOS as Internet provider?

Not sure what i am missing but traffic dies at the Fios router (Actiontec) Sip provider/NexVortex

7 Replies 7

priyk
Level 1
Level 1

Hi ,

Could you please elaborate, what do you mean by traffic dies??

priyk
Level 1
Level 1

Hi ,

Could you please elaborate, what do you mean by traffic dies??

I need help setting up Fios router for port forwarding to allow signalling udp 5060 and rtp 10000-20000. I thought i did but i still can't make outbound call. Inbound calls are not coming in either....

Hi,

Do you hear fast busy when you mean  incoming/outgoing calls not working?.What is the complete set up ..any firewall ?.callmanager traces would be helpful to figure out whats going on..

I even opened a case with Cisco TAC and config is good the issue is the fios router it does have fw in it. Provider is not seeing any traffic from me basically

Correct , TAC wont be able to troubleshoot on third party device.configuration can be checked and with traces we can  make sure if  we are sending the traffic out through the trunk. Unfortunate that i couldnt help..To answer to your first question, i have never set up SIP trunk to FIOS..sorry!:)

voiplab-rt1#
*May 22 17:12:31.514: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:12404224588@66.23.129.253:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.254:5060;branch=z9hG4bK6455DA
From: "deleted" <2071>;tag=524944C-8A9
To: <12404224588>
Date: Sun, 22 May 2011 17:12:31 GMT
Call-ID: 8346BBEB-83CD11E0-801D9E8B-E8DABB46@nexvortex.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2178331263-2211254752-2149097099-3906648902
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1306084351
Contact: <2071>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 328

v=0
o=CiscoSystemsSIP-GW-UserAgent 4758 1976 IN IP4 10.10.0.254
s=SIP Call
c=IN IP4 10.10.0.254
t=0 0
m=audio 19042 RTP/AVP 0 18 100 101
c=IN IP4 10.10.0.254
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

*May 22 17:12:32.014: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:12404224588@66.23.129.253:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.254:5060;branch=z9hG4bK6455DA
From: "deleted" <2071>;tag=524944C-8A9
To: <12404224588>
Date: Sun, 22 May 2011 17:12:32 GMT
Call-ID: 8346BBEB-83CD11E0-801D9E8B-E8DABB46@nexvortex.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2178331263-2211254752-2149097099-3906648902
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1306084352
Contact: <2071>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 328

v=0
o=CiscoSystemsSIP-GW-UserAgent 4758 1976 IN IP4 10.10.0.254
s=SIP Call
c=IN IP4 10.10.0.254
t=0 0
m=audio 19042 RTP/AVP 0 18 100 101
c=IN IP4 10.10.0.254
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

*May 22 17:12:33.014: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:12404224588@66.23.129.253:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.254:5060;branch=z9hG4bK6455DA
From: "deleted" <2071>;tag=524944C-8A9
To: <12404224588>
Date: Sun, 22 May 2011 17:12:33 GMT
Call-ID: 8346BBEB-83CD11E0-801D9E8B-E8DABB46@nexvortex.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2178331263-2211254752-2149097099-3906648902
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1306084353
Contact: <2071>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 328

v=0
o=CiscoSystemsSIP-GW-UserAgent 4758 1976 IN IP4 10.10.0.254
s=SIP Call
c=IN IP4 10.10.0.254
t=0 0
m=audio 19042 RTP/AVP 0 18 100 101
c=IN IP4 10.10.0.254
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


voiplab-rt1#
*May 22 17:10:42.998: //1605/3CDC01128012/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4A807270
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 2071
Called Number            : 12404224588
Source IP Address (Sig  ): 10.10.0.254
Destn SIP Req Addr:Port  : 66.23.129.253:5060
Destn SIP Resp Addr:Port : 66.23.129.253:5060
Destination Name         : 66.23.129.253

*May 22 17:10:42.998: //1605/3CDC01128012/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec  
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 10.10.0.254
Source IP Port    (Media): 18402
Destn  IP Address (Media):  -
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

*May 22 17:10:42.998: //1605/3CDC01128012/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 102
Disconnect Cause (SIP)   : 200

Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: