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Help with Verizon dial config in Australia

dsieme01
Level 1
Level 1

‎05-15-2019 11:17 AM

I setting up new service in Austrialia  with a cube and Verizon.  I can take incoming calls but outgoing calls are failing.   I've tried with other codec options  

 

my dialpeer is simple  

dial-peer voice 2000 voip
 description Outound for Verizon Voip Service
 destination-pattern ......+
 session protocol sipv2
 session target ipv4:172.30.201.11
 voice-class sip options-keepalive up-interval 100 down-interval 50 retry 6
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

 

 

Any help would be appreciated. 

 

 

My messages and error debug on.     

 

 

aumgw21#debug ccsip error
SIP Call error tracing is enabled
aumgw21#

May 15 18:11:10.385: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:001119132676897@10.117.220.15 SIP/2.0
Via: SIP/2.0/UDP 10.117.220.11:5060;branch=z9hG4bK0096c49e
Max-Forwards: 70
From: "Daniel Siemens" <sip:32297@10.117.220.11>;tag=as57ac376d
To: <sip:001119132676897@10.117.220.15>
Contact: <sip:32297@10.117.220.11:5060>
Call-ID: 69b95f1468b32a8b0275e8c21634d204@10.117.220.11:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.25.1
Date: Wed, 15 May 2019 18:11:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Daniel Siemens" <sip:32297@10.117.220.11>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 1252680437 1252680437 IN IP4 10.117.220.11
s=Asterisk PBX 11.25.1
c=IN IP4 10.117.220.11
t=0 0
m=audio 11232 RTP/AVP 0 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
May 15 18:11:10.385: //-1/xxxxxxxxxxxx/SIP/Error/httpish_msg_free:
 Freeing NULL pointer!
SIP: (536518) Attribute mid, level 1 instance 1 not found.
SIP: (536518) setup attribute, level 1 instance 1 not found.
SIP: (536518) connection attribute, level 1 instance 1 not found.
SIP: (536518) Attribute label, level 1 instance 1 not found.
SIP: (536518) a=framerate attribute, level 1 instance 1 not found.
May 15 18:11:10.393: //536519/A6D5CD15B0E9/SIP/Error/sipSPIGetCallServerGroupTargets:
 No server group configured
May 15 18:11:10.393: //-1/xxxxxxxxxxxx/SIP/Error/voipCodec_to_rtpAvpCodec:
 Unexpected VoIPCodec Type :No Codec  
May 15 18:11:10.393: //-1/xxxxxxxxxxxx/SIP/Error/voipCodec_to_rtpAvpCodec:
 Unexpected VoIPCodec Type :No Codec  
May 15 18:11:10.393: //-1/xxxxxxxxxxxx/SIP/Error/voipCodec_to_rtpAvpCodec:
 Unexpected VoIPCodec Type :No Codec  
SIP: (536519) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: (536519) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: (536519) Group (a= group line) attribute, level 65535 instance 1 not found.
May 15 18:11:10.397: //536518/A6D5CD15B0E9/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.117.220.11:5060;branch=z9hG4bK0096c49e
From: "Daniel Siemens" <sip:32297@10.117.220.11>;tag=as57ac376d
To: <sip:001119132676897@10.117.220.15>
Date: Thu, 16 May 2019 04:11:10 GMT
Call-ID: 69b95f1468b32a8b0275e8c21634d204@10.117.220.11:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M9
Content-Length: 0

May 15 18:11:10.397: //536519/A6D5CD15B0E9/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:001119132676897@172.30.201.11:5060 SIP/2.0
Via: SIP/2.0/UDP 10.117.220.15:5060;branch=z9hG4bK528B52B5
From: "Daniel Siemens" <sip:32297@10.117.220.15>;tag=A94BB25C-664
To: <sip:001119132676897@172.30.201.11>
Date: Thu, 16 May 2019 04:11:10 GMT
Call-ID: A6D70565-767311E9-B0EFA9B3-D26A7C2C@10.117.220.15
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2799029525-1987252713-2968103347-3530193964
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M9
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1557943870
Contact: <sip:32297@10.117.220.15:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
P-Asserted-Identity: "Daniel Siemens" <sip:32297@10.117.220.15>
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 3623 6230 IN IP4 10.117.220.15
s=SIP Call
c=IN IP4 10.117.220.15
t=0 0
m=audio 16612 RTP/AVP 0 101
c=IN IP4 10.117.220.15
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
May 15 18:11:10.417: //-1/xxxxxxxxxxxx/SIP/Error/get_content_length:
 Could not get Content-length
May 15 18:11:10.421: //536519/A6D5CD15B0E9/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.117.220.15:5060;branch=z9hG4bK528B52B5
From: "Daniel Siemens" <sip:32297@10.117.220.15>;tag=A94BB25C-664
To: <sip:001119132676897@172.30.201.11>
Call-ID: A6D70565-767311E9-B0EFA9B3-D26A7C2C@10.117.220.15
CSeq: 101 INVITE
Timestamp: 1557943870

May 15 18:11:10.689: //536519/A6D5CD15B0E9/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.117.220.15:5060;branch=z9hG4bK528B52B5
From: "Daniel Siemens" <sip:32297@10.117.220.15>;tag=A94BB25C-664
To: <sip:001119132676897@172.30.201.11>;tag=1488613558-1557943870704
Call-ID: A6D70565-767311E9-B0EFA9B3-D26A7C2C@10.117.220.15
CSeq: 101 INVITE
Timestamp: 1557943870
Supported:
Contact: <sip:001119132676897@172.30.201.11:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Length: 0

May 15 18:11:10.689: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_reg_search_reg_number_table:
 No entry found in reg Number Table for 32297
May 15 18:11:10.689: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_reg_search_reg_number_table:
 No entry found in reg Number Table for 001119132676897
May 15 18:11:10.689: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr:
 ReqLine IP addr does not match with host IP addr
May 15 18:11:10.693: //536518/A6D5CD15B0E9/SIP/Msg/ccsipDisplayMsg:
Sent:
aumgw21#SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.117.220.11:5060;branch=z9hG4bK0096c49e
From: "Daniel Siemens" <sip:32297@10.117.220.11>;tag=as57ac376d
To: <sip:001119132676897@10.117.220.15>;tag=A94BB384-128C
Date: Thu, 16 May 2019 04:11:10 GMT
Call-ID: 69b95f1468b32a8b0275e8c21634d204@10.117.220.11:5060
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:001119132676897@10.117.220.15>;party=called;screen=no;privacy=off
Contact: <sip:001119132676897@10.117.220.15:5060>
Server: Cisco-SIPGateway/IOS-15.4.3.M9
Content-Length: 0

May 15 18:11:11.197: //536519/A6D5CD15B0E9/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.117.220.15:5060;branch=z9hG4bK528B52B5
From: "Daniel Siemens" <sip:32297@10.117.220.15>;tag=A94BB25C-664
To: <sip:001119132676897@172.30.201.11>;tag=1488613558-1557943870704
Call-ID: A6D70565-767311E9-B0EFA9B3-D26A7C2C@10.117.220.15
CSeq: 101 INVITE
Timestamp: 1557943870
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp
Contact: <sip:001119132676897@172.30.201.11:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 228
v=0
o=BroadWorks 9916054 1 IN IP4 172.30.201.132
s=-
c=IN IP4 172.30.201.132
t=0 0
m=audio 10080 RTP/AVP 0 101
c=IN IP4 172.30.201.132
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
SIP: Attribute mid, level 1 instance 1 not found.
SIP: setup attribute, level 1 instance 1 not found.
SIP: connection attribute, level 1 instance 1 not found.
SIP: Attribute label, level 1 instance 1 not found.
SIP: a=framerate attribute, level 1 instance 1 not found.
May 15 18:11:11.201: //-1/xxxxxxxxxxxx/SIP/Error/voipCodec_to_rtpAvpCodec:
 Unexpected VoIPCodec Type :No Codec  
May 15 18:11:11.201: //536519/A6D5CD15B0E9/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:001119132676897@172.30.201.11:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.117.220.15:5060;branch=z9hG4bK528B61383
From: "Daniel Siemens" <sip:32297@10.117.220.15>;tag=A94BB25C-664
To: <sip:001119132676897@172.30.201.11>;tag=1488613558-1557943870704
Date: Thu, 16 May 2019 04:11:10 GMT
Call-ID: A6D70565-767311E9-B0EFA9B3-D26A7C2C@10.117.220.15
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

SIP: (536518) Group (a= group line) attribute, level 65535 instance 1 not found.
May 15 18:11:11.205: //536518/A6D5CD15B0E9/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.117.220.11:5060;branch=z9hG4bK0096c49e
From: "Daniel Siemens" <sip:32297@10.117.220.11>;tag=as57ac376d
To: <sip:001119132676897@10.117.220.15>;tag=A94BB384-128C
Date: Thu, 16 May 2019 04:11:10 GMT
Call-ID: 69b95f1468b32a8b0275e8c21634d204@10.117.220.11:5060
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:001119132676897@10.117.220.15>;party=called;screen=no;privacy=off
Contact: <sip:001119132676897@10.117.220.15:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.4.3.M9
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 3323 9341 IN IP4 10.117.220.15
s=SIP Call
c=IN IP4 10.117.220.15
t=0 0
m=audio 16610 RTP/AVP 0 101
c=IN IP4 10.117.220.15
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
May 15 18:11:11.209: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
aumgw21#ACK sip:001119132676897@10.117.220.15:5060 SIP/2.0
Via: SIP/2.0/UDP 10.117.220.11:5060;branch=z9hG4bK70a56a90
Max-Forwards: 70
From: "Daniel Siemens" <sip:32297@10.117.220.11>;tag=as57ac376d
To: <sip:001119132676897@10.117.220.15>;tag=A94BB384-128C
Contact: <sip:32297@10.117.220.11:5060>
Call-ID: 69b95f1468b32a8b0275e8c21634d204@10.117.220.11:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.25.1
Content-Length: 0

aumgw21#
May 15 18:11:13.957: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:001119132676897@10.117.220.15:5060 SIP/2.0
Via: SIP/2.0/UDP 10.117.220.11:5060;branch=z9hG4bK0b17d9f7
Max-Forwards: 70
From: "Daniel Siemens" <sip:32297@10.117.220.11>;tag=as57ac376d
To: <sip:001119132676897@10.117.220.15>;tag=A94BB384-128C
Call-ID: 69b95f1468b32a8b0275e8c21634d204@10.117.220.11:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.25.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

May 15 18:11:13.961: //536518/A6D5CD15B0E9/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.117.220.11:5060;branch=z9hG4bK0b17d9f7
From: "Daniel Siemens" <sip:32297@10.117.220.11>;tag=as57ac376d
To: <sip:001119132676897@10.117.220.15>;tag=A94BB384-128C
Date: Thu, 16 May 2019 04:11:13 GMT
Call-ID: 69b95f1468b32a8b0275e8c21634d204@10.117.220.11:5060
Server: Cisco-SIPGateway/IOS-15.4.3.M9
CSeq: 103 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=127,OS=20320,PR=123,OR=19680,PL=0,JI=0,LA=0,DU=2
Content-Length: 0

May 15 18:11:13.961: //536519/A6D5CD15B0E9/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:001119132676897@172.30.201.11:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.117.220.15:5060;branch=z9hG4bK528B7697
From: "Daniel Siemens" <sip:32297@10.117.220.15>;tag=A94BB25C-664
To: <sip:001119132676897@172.30.201.11>;tag=1488613558-1557943870704
Date: Thu, 16 May 2019 04:11:10 GMT
Call-ID: A6D70565-767311E9-B0EFA9B3-D26A7C2C@10.117.220.15
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M9
Max-Forwards: 70
P-Asserted-Identity: "Daniel Siemens" <sip:32297@10.117.220.15>
Timestamp: 1557943873
CSeq: 102 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=123,OS=19680,PR=127,OR=20320,PL=0,JI=0,LA=0,DU=2
Content-Length: 0

aumgw21#
May 15 18:11:13.985: //536519/A6D5CD15B0E9/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.117.220.15:5060;branch=z9hG4bK528B7697
From: "Daniel Siemens" <sip:32297@10.117.220.15>;tag=A94BB25C-664
To: <sip:001119132676897@172.30.201.11>;tag=1488613558-1557943870704
Call-ID: A6D70565-767311E9-B0EFA9B3-D26A7C2C@10.117.220.15
Timestamp: 1557943873
CSeq: 102 BYE
Content-Length: 0
Labels:
0 Helpful
1 Reply 1

Giuseppe Larosa
Hall of Fame
Hall of Fame

‎05-16-2019 05:05 AM

Hello Daniel,

I would suggest to move your thread to IP telephony forum to get better help.

 

Best Regards

Giuseppe

 

0 Helpful
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