SIP IOS Gateway Outbound Calls Continue to Ring

[ashley.philip]

I am trying to resolve an issue wherein the outogoing call continues to ring even after the end user has anwsered the phone.  But the incoming call works fine.  This is an IOS Gateway that is connected to a SIP provider and the call is made over the SIP trunk.  Any help in resolving this would be appreciated.  Let me know if I can provide you debugs or other information. 

I am not seeing any issues with dial-peers as the call is going out and hence I have not copied them here.  However let me know if you need this.  Thanks

Ashley

[Ayodeji Okanlawon]

Ashley,

Can you please attach a sh run of your router.

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[ashley.philip]

Thanks for taking the time to review this issue.  I have copied the configuration below.

[ashley.philip]

Hi ,

Configuration is copied below.

Thanks for taking the time.

=====

multilink bundle-name authenticated

!

voice-card 0

no dspfarm

!

!

!

voice call send-alert

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

sip

  no call service stop

!

!

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g729r8

!

voice translation-rule 1

rule 1 /^2169\(3..\)$/ /7\1/

!

voice translation-rule 2

rule 1 /^7\(3..\)$/ /012169\1/

!

voice translation-rule 3

rule 1 /^7\(3..\)$/ /2169\1/

!

!

voice translation-profile EXT

translate called 1

!

voice translation-profile INTERNATIONAL

translate calling 3

!

voice translation-profile LOCAL

translate calling 2

!

!

archive

log config

  hidekeys

!

!

interface GigabitEthernet0/0

ip address 10.66.7.218 255.255.255.252

ip virtual-reassembly

duplex auto

speed auto

media-type rj45

!

interface GigabitEthernet0/1

ip address 10.106.78.7 255.255.255.128

ip virtual-reassembly

duplex auto

speed auto

media-type rj45

!

ip route 0.0.0.0 0.0.0.0 10.106.78.1

ip route 10.200.7.156 255.255.255.252 10.66.7.217

!

!

ip http server

no ip http secure-server

ip nat source list 1 interface GigabitEthernet0/0

!

access-list 1 permit 10.106.78.7

!

!

!

!

!

!

control-plane

!

!

!

voice-port 0/1/0

!

voice-port 0/1/1

!

voice-port 0/1/2

!

voice-port 0/1/3

!

!

!

!

!        

dial-peer voice 7300 voip

translation-profile outgoing EXT

destination-pattern 21693..

session protocol sipv2

session target ipv4:10.104.160.29

incoming called-number .

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 7301 voip

translation-profile outgoing EXT

preference 2

destination-pattern 21693..

session protocol sipv2

session target ipv4:10.104.160.30

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 2030 voip

description Test

destination-pattern 00T

session protocol sipv2

session target ipv4:10.200.7.157

session transport udp

incoming called-number 21693..

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 2040 voip

description National Numbers

destination-pattern 0........

session protocol sipv2

session target ipv4:10.200.7.157

session transport udp

dtmf-relay rtp-nte

codec g711alaw

!

!

sip-ua

[Ayodeji Okanlawon]

Thanks Ashley. Please do a test call and send the ff;

1. debug ccsip messages

2. debug voip ccapi inout

please put the logs in a text file and attach here. Dont paste them directly as it makes the thread very long. Thanks

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[ashley.philip]

Hi

files are attached.

[Ayodeji Okanlawon · Accepted solution]

Ashley after looking at your trace, I can see whats going on. Your provider is not sending 200 okay..

Here in your trace your provider sends 180 with sdp but after that they never sent a 200 ok. You need to contact them. If you want a bullet proof reason that it is their problem you can send me a trace for incoming call and I will show you that cucm sends a 200 ok for the call. You can then put this nicely at their feet and make them earn their wages

+++First 180 with sdp received from telco++++++++++

Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.66.7.218:5060;branch=z9hG4bK6B81
Record-Route: <10.200.7.157:5060>
Call-ID: C5F1CD73-CB5E11E1-84FAAB7B-E412AAC8@10.66.7.218
From: "Riyadh Reception"<2169300>;tag=FD8AC4-82C
To: <0097339982779>;tag=sbc0806dd2cbk7s-CC-36
CSeq: 101 INVITE
Contact: <0097339982779>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Require: 100rel
RSeq: 1
Content-Length: 369
Content-Type: application/sdp

v=0
o=- 6268406 6268406 IN IP4 10.200.7.157
s=SBC call
c=IN IP4 10.200.7.157
t=0 0
m=audio 43398 RTP/AVP 8 0 18 4 2 98 99 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

*Jul 12 13:46:18.302: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

+++second 180 with sdp received from telco++++++++++

Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.66.7.218:5060;branch=z9hG4bK6B81
Record-Route: <10.200.7.157:5060>
Call-ID: C5F1CD73-CB5E11E1-84FAAB7B-E412AAC8@10.66.7.218
From: "Riyadh Reception"<2169300>;tag=FD8AC4-82C
To: <0097339982779>;tag=sbc0806dd2cbk7s-CC-36
CSeq: 101 INVITE
Contact: <0097339982779>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Require: 100rel
RSeq: 1
Content-Length: 369
Content-Type: application/sdp

v=0
o=- 6268406 6268406 IN IP4 10.200.7.157
s=SBC call
c=IN IP4 10.200.7.157
t=0 0
m=audio 43398 RTP/AVP 8 0 18 4 2 98 99 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

*Jul 12 13:46:18.958: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.66.7.218:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKukebhftds22bachpfu72da2ft
Call-ID: isbcudfu4774ebcd24dspaffffuefo2d2k2o@SoftX3000
From: <10.66.7.218:5060>;tag=sbc0807beaosoop
To: <10.66.7.218>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

+++Third 180 with sdp received from telco++++++++++

*Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.66.7.218:5060;branch=z9hG4bK6B81
Record-Route: <10.200.7.157:5060>
Call-ID: C5F1CD73-CB5E11E1-84FAAB7B-E412AAC8@10.66.7.218
From: "Riyadh Reception"<2169300>;tag=FD8AC4-82C
To: <0097339982779>;tag=sbc0806dd2cbk7s-CC-36
CSeq: 101 INVITE
Contact: <0097339982779>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Require: 100rel
RSeq: 1
Content-Length: 369
Content-Type: application/sdp

v=0
o=- 6268406 6268406 IN IP4 10.200.7.157
s=SBC call
c=IN IP4 10.200.7.157
t=0 0
m=audio 43398 RTP/AVP 8 0 18 4 2 98 99 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

*Jul 12 13:46:24.302: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

+++Fourth 180 with sdp received from telco++++++++++

Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.66.7.218:5060;branch=z9hG4bK6B81
Record-Route: <10.200.7.157:5060>
Call-ID: C5F1CD73-CB5E11E1-84FAAB7B-E412AAC8@10.66.7.218
From: "Riyadh Reception"<2169300>;tag=FD8AC4-82C
To: <0097339982779>;tag=sbc0806dd2cbk7s-CC-36
CSeq: 101 INVITE
Contact: <0097339982779>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Require: 100rel
RSeq: 1
Content-Length: 369
Content-Type: application/sdp

v=0
o=- 6268406 6268406 IN IP4 10.200.7.157
s=SBC call
c=IN IP4 10.200.7.157
t=0 0
m=audio 43398 RTP/AVP 8 0 18 4 2 98 99 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

*Jul 12 13:46:32.298: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

+++Fifth 180 with sdp received from telco++++++++++

Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.66.7.218:5060;branch=z9hG4bK6B81
Record-Route: <10.200.7.157:5060>
Call-ID: C5F1CD73-CB5E11E1-84FAAB7B-E412AAC8@10.66.7.218
From: "Riyadh Reception"<2169300>;tag=FD8AC4-82C
To: <0097339982779>;tag=sbc0806dd2cbk7s-CC-36
CSeq: 101 INVITE
Contact: <0097339982779>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Require: 100rel
RSeq: 1
Content-Length: 369
Content-Type: application/sdp

v=0
o=- 6268406 6268406 IN IP4 10.200.7.157
s=SBC call
c=IN IP4 10.200.7.157
t=0 0
m=audio 43398 RTP/AVP 8 0 18 4 2 98 99 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

Ideally after the first 180 ringing your provider should send a 200 ok. I can bet this is Huwaei!

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[ashley.philip]

Thank you very much for this insight.  I was able to compare the logs and see the issue.  Thank you once again