11-20-2013 11:04 PM - edited 03-16-2019 08:31 PM
Hi guys,could any one help me in the following.
ITSP-->Voice gateway configured as CUBE-->CUCM-->UCCX
I am moving a system from cme and aa enviroment to cucm and uccx
The VGW is configured as CUBE and also is added as h323 gateway on cucm.
When i tested the debug ccsip messages shows
Sip 503 service unavailable or
sip 500 internal server error.
I can't now provide any debugs cause i am not on site,only on Saturday.
As i read in previous discussion that could be the bind source address problem but i had this configured.
Also i tried to configure the gateway instead of h232 to use sip trunk from cucm,but after this the incoming calls didn't even reach the router,the debug ccsip messages showed nothing.
For now can any one advice me to what these 2 errors related to.
What could be missing?
Thanks in advance.
11-21-2013 12:17 AM
Ahmed,
Its difficult to say unless we see both configs and the logs...Usually when you get a 5XX error, you should see the reason code for that error...Can you share that here. When you can send the ff:
1. sh run
2. debug voip ccapi inout
3.debug ccsip messages
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
11-21-2013 12:26 AM
Hi Aokanlawon,
Actually i can't now ,so kindly asking you to check this post on Saturday and i ll update you with all the configs and debugs.
Thanks man.
11-23-2013 12:35 AM
Hi there : can some one explain the reason that i am getting this sip error with itsp:
here is the debug of ccsip messages:
Received:
INVITE sip:5555317884@178.208.X.X;user=phone SIP/2.0
Via: SIP/2.0/UDP 188.254.68.66:9298;branch=z9hG4bK-6110d60075a24c0f-a3c000c-1
Call-ID: isbc6994325518768294927-1385194135-11717
From: <>>9268854639@sipgw120.com;user=phone>;tag=sbc09106994325518768294927
To: <5555317884>5555317884>
CSeq: 1 INVITE
Min-SE: 90
Session-Expires: 3600;refresher=uac
Contact: <9268854936>9268854936>
Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO,PRACK
Supported: timer,100rel
Diversion: <>>8002001706@sipgw120.com>;privacy=off;screen=no;reason=unknown,<>>8002001706@sipgw120.com>;privacy=off;screen=no;reason=unknown
Max-Forwards: 70
User-Agent: VCS 5.8.2.56-03
Content-Length: 394
Content-Type: application/sdp
v=0
o=- 87852 198805 IN IP4 188.254.68.67
s=SBC call
c=IN IP4 188.254.68.67
t=0 0
m=audio 23682 RTP/AVP 8 0 18 98 96 97 101
a=rtpmap:98 G.729a/8000
a=rtpmap:96 G.729ab/8000
a=rtpmap:97 G.729b/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=ptime:10
a=X-vrzcap:vbd Ver=1 Mode=FaxPr ModemRtpRed=0
a=X-vrzcap:identification bin=DSR2866 Prot=mgcp App=MG
00:43:23: //11/FDB448CE8020/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 188.254.68.66:9298;branch=z9hG4bK-6110d60075a24c0f-a3c000c-1
From: <>>9268854639@sipgw120.com;user=phone>;tag=sbc09106994325518768294927
To: <5555317884>5555317884>
Date: Sat, 23 Nov 2013 08:06:29 GMT
Call-ID: isbc6994325518768294927-1385194135-11717
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
00:43:23: //11/FDB448CE8020/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 188.254.68.66:9298;branch=z9hG4bK-6110d60075a24c0f-a3c000c-1
From: <>>9268854936@sipgw120.com;user=phone>;tag=sbc09106994325518768294927
To: <5555317884>5555317884>
c2801#er=phone>;tag=27BA64-1DAE
Date: Sat, 23 Nov 2013 08:06:29 GMT
Call-ID: isbc6994325518768294927-1385194135-11717
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=38
Content-Length: 0
00:43:23: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:5555317884@178.208.129.221;user=phone SIP/2.0
Via: SIP/2.0/UDP 188.254.68.66:9298;branch=z9hG4bK-6110d60075a24c0f-a3c000c-1
Call-ID: isbc6994325518768294927-1385194135-11717
From: <>>9268854639@sipgw120.com;user=phone>;tag=sbc09106994325518768294927
To: <5555317884>;tag=27BA64-1DAE5555317884>
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
show run:
voice service voip
ip address trusted list
ipv4 87.226.136.164 255.255.255.255
ipv4 172.16.24.0 255.255.255.0
ipv4 188.254.68.66 255.255.255.255
ipv4 188.254.68.67 255.255.255.255
ipv4 188.254.69.66 255.255.255.255
ipv4 188.254.69.67 255.255.255.255
ipv4 46.38.52.68 255.255.255.255
address-hiding
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco
sip
voice class codec 1
codec preference 1 g729br8
codec preference 2 g729r8
codec preference 3 g711alaw
codec preference 4 g711ulaw
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 1
rule 1 /XXX5397962/ /1999/
!
voice translation-rule 2
rule 1 /XXX55317577/ /1999/
!
voice translation-rule 3
rule 1 /5555317884/ /1999/
!
!
voice translation-profile ROS
translate called 1
!
voice translation-profile ROS2
translate called 2
!
voice translation-profile ROS3
translate called 3
interface FastEthernet0/0
ip address 178.208.129.221 255.255.255.248
ip access-group INBOUND in
no ip unreachables
ip verify unicast reverse-path
ip nat outside
ip inspect IPFW in
ip inspect IPFW out
ip virtual-reassembly in
duplex auto
speed auto
no cdp enable
!
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly in
duplex auto
speed auto
!
interface FastEthernet0/1.1
encapsulation dot1Q 1 native
ip address 10.110.0.200 255.255.255.0
ip nat inside
ip virtual-reassembly in
!
interface FastEthernet0/1.2
encapsulation dot1Q 2
ip address 172.16.24.254 255.255.255.0
ip nat inside
ip virtual-reassembly in
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.24.254
!
ip dns server
ip nat inside source list NAT interface FastEthernet0/0 overload
ip route 0.0.0.0 0.0.0.0 178.208.X.X
ip route 192.168.0.0 255.255.0.0 Null0 254
sccp local FastEthernet0/1.2
sccp ccm 172.16.24.101 identifier 1 version 7.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register XCODE123456
keepalive retries 1
keepalive timeout 10
switchover method immediate
switchback method immediate
!
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 6
associate application SCCP
!
dial-peer voice 10000 voip
tone ringback alert-no-PI
description ROSTELECOM Incoming
translation-profile incoming ROS
destination-pattern 74955397962
session protocol sipv2
session target ipv4:87.226.136.164
session transport udp
incoming called-number XXXX5397962
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 10010 voip
tone ringback alert-no-PI
description ROSTELECOM Incoming
translation-profile incoming ROS2
destination-pattern XXX55317577
session protocol sipv2
session target ipv4:87.226.136.164
session transport udp
incoming called-number 75555317577
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 10020 voip
tone ringback alert-no-PI
description ROSTELECOM Incoming
translation-profile incoming ROS3
preference 1
destination-pattern 5555317884
session protocol sipv2
session target ipv4:188.254.68.66
session transport udp
incoming called-number 5555317884
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 10021 voip
tone ringback alert-no-PI
description ROSTELECOM Incoming
translation-profile incoming ROS
preference 2
destination-pattern 5555317884
session protocol sipv2
session target ipv4:188.254.69.66
session transport udp
incoming called-number 5555317884
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 2 voip
tone ringback alert-no-PI
description to CUCM_PUB
destination-pattern 1...
session target ipv4:172.16.24.101
voice-class codec 2
dtmf-relay rtp-nte
******************************************
I see in the debug that the itsp over g729 family codecs but not g711 at all
This system was working with this dialpeers before with same provider ,just i have added the dial-peer 2 .
I have changed the codec to match what is offered by itsp but no difference,still getting the same message.
PLZ help ASAP.
11-24-2013 03:16 AM
Hi Ahmed,
As we see that the gateway is sending 503 Service Unavailable and it is communicating over H323 with the CUCM, please provide the following debugs as well with the 'deb ccsip messages' to isolate if the disconnect is being received from the other leg or by the gateway itself.
1. debug voip ccapi inout
2. debug h225 asn1
3. debug h245 asn1
Also, we see that the SDP in the invite from the provider supports G.729a, G.729ab and G.729b only. The dial peers for the SIP leg is hardcoded with G711ulaw only.
Please try to configure a new voice class codec for support of G.729a, G.729ab and G.729b and G711ulaw. Then, add it to the inbound dial peers (SIP leg).
HTH,
Jagpreet Singh Barmi
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