H323 and SIP Interworking

Any idea which Gateway can support H323 and SIP interworking(RFC4123)?

Hi,
Cisco IPIPGW IOS can support h323 and Sip interworking.
You can find it under voice products.

Similar Messages

H323 and SIP

I currently have an H323 gateway with one PRI. All the phones are SCCP phones registering to CUCM. I want to add a 9971 SIP phone to this gateway. The only configuration I have done thus far is to allow connection between H323 and SIP, SIP to H323. The internal calls work fine, but when trying to dial out there is just blank noise and it eventually goes busy and disconnects. Can anyone help me set this up or point me in the right direction.
Thanks!!

Hi,
Cisco IPIPGW IOS can support h323 and Sip interworking.
You can find it under voice products.

Cucm 10.1version - training videos and hand guides on understanding voice gateway h323 and SIP? thanks

cucm 10.1version - any free training videos and hand guides on understanding voice gateway h323 and SIP and how to configure one? thanks

Learncisco gives a very good introduction to CUCM - I recommend you start there.

Running H323 and SIP on the same cisco gateway

Hi there!
I running Yate and Asterisk/IAX2 in the same box, to get H323 and convert to SIP, but I have a costumer that wants to send H323 with codec g723, but it doesn?t work good at all. So, can have H323 and SIP running on the same box like a c3600 or c2600?
Thanks,
Jonas

I may go out of topic, but:
1. If you want reliable protocol (H323<-> SIP) and codec conversation and
2. If you wanto to show your cusotomer nice PBX features,
then go fo either MERA MVTS or Alterteks PSS softswitches.

AS5400 Performance runining Both H323 and SIP

Dear All,
Is there any way to run Voice Gateway like AS5400 with two protocol H323 and SIP simultaneously? Any voice gateway performance afftected? or Voice quality affected? if we run both protocol in only one gateway?
Best Regards,
Daneth

AS5400 supports H.323 and SIP dial-peer at the same time without problems.
I've used AS5400 in IP2IP gateway mode to convert SIP in H.323 and vice versa with about 150 concurrent calls.
In lab I also tested SIP, H.323 and MGCP at same time.
In default configuration SIP and H.323 are both active.
AS5400 uses H.323 like default signalling protocol. Is sufficient create a voip dial-peer. To specify SIP you must use the command "session protocol sipv2" under a dial-peer.
To shut down SIP use
voice service voip
sip
call service stop
To shut down H.323 use
no gateway

Why we dont' see H323 gateway and SIP trunk "registered" in CCM?

in CCM admin, we see the status of H323 gateways and SIP trunks have no "registered to xxxx" status.

Unlike devices with other protocols, there is no registration mechanism for these.

H323 as CUBE and SIP Gateway

I have a H.323 gateway. I want to know if I can configure this existing h.323 gateway as a Cube to transport SIP.
If so do I need to bind a seperate interface to SIP and H.323 or the same interface can be used for both
Can the Gateway be registered as h.323 to CCM1 and Can I have a SIP trunk between the same Gateway to CCM2?
Thanks in advance

Hi,
"I have a H.323 gateway. I want to know if I can configure this existing h.323 gateway as a Cube to transport SIP".
A CUBE gateway by definition is a voice gateway that connects ip-2-ip calls and plus some extra fancy features. The gateway can support both h.323 and sip at the same time.
"If so do I need to bind a separate interface to SIP and H.323 or the same interface can be used for both"
This is tricky question, the answer is I don't know but why limit yourself when you can bind them into different loopback interfaces.
"Can the Gateway be registered as h.323 to CCM1 and Can I have a SIP trunk between the same Gateway to CCM2?"
Absolutely.

Cucm , Cube via Sip and Sip Trunk to ISP , Outgoing calls not working

Hi
We have issue with the outgoing calls to sip trunk
Below is the config and the debugs
It will be great if you give your thoughts since we have stuck here
My thoughts are:
i see that for unknown reason the called number is going with 4 digits instead of 8 digits
i dont see any sip message comming from ISP
Maybe the call not going there ? to isp trunk? From the trace the call hit the correct dialpeer 888 but i see 4 digits as a called number , but i dodnt understant the reason to translated in 4 digits the called number.Not apply a translation rule for that
confused!!!
Calling Numbner:22324086
Called Number: 23823690
CUCM:192.168.1.241 and 242
CUBE:192.168.1.10
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
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 h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*$....$/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^$9$$.......$$/ /99\2/
rule 3 /^$2$$.......$$/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
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 h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*$....$/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^$9$$.......$$/ /99\2/
rule 3 /^$2$$.......$$/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad

Hi Aok
I change the default value for IPVMS from g711ulaw to g711alaw but the results remained the same
Also i have restarted the IPVMS
SIP-GW#
SIP-GW#
*Mar 5 14:19:57.854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 38874 2 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24784 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:57.878: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1362493197
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3807 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 101
c=IN IP4 10.249.13.130
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.878: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:57.926: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 102 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 213
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 2 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=inactive
*Mar 5 14:19:57.942: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5774 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 101
c=IN IP4 192.168.1.10
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.946: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3562A4
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:57.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK798246ab3597
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
*Mar 5 14:19:58.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Length: 0
*Mar 5 14:19:58.158: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1362493198
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:58.158: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:58.218: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 103 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 216
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 3 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 18 96 99
a=rtpmap:96 AMR/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
a=sendrecv
*Mar 5 14:19:58.234: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 283
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5775 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 18 101
c=IN IP4 192.168.1.10
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:58.242: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7985648033f2
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 192
v=0
o=CiscoSystemsCCM-SIP 38874 3 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 192.168.1.241
t=0 0
m=audio 4000 RTP/AVP 8
a=X-cisco-media:umoh
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly
*Mar 5 14:19:58.262: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK358582
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3808 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 99
c=IN IP4 10.249.13.130
a=sendonly
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
SIP-GW#
SIP-GW#sh voip rtp connections
VoIP RTP active connections :
No. CallId     dstCallId LocalRTP RmtRTP     LocalIP                                RemoteIP
1     716        717        19314    4000     192.168.1.10                           192.168.1.241
2     717        716        19234    54932    10.249.13.130                          10.224.42.72
Found 2 active RTP connections

Sip 503 service unavailable and sip 500 internal server error

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.

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:[email protected];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: [email protected];user=phone>;tag=sbc09106994325518768294927
To:
CSeq: 1 INVITE
Min-SE: 90
Session-Expires: 3600;refresher=uac
Contact:
Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO,PRACK
Supported: timer,100rel
Diversion: [email protected]>;privacy=off;screen=no;reason=unknown,[email protected]>;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: [email protected];user=phone>;tag=sbc09106994325518768294927
To:
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: [email protected];user=phone>;tag=sbc09106994325518768294927
To:
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:[email protected];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: [email protected];user=phone>;tag=sbc09106994325518768294927
To: ;tag=27BA64-1DAE
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.

How to configure modem connection with GW (H323) and ATA 187

Hello Community,
i stock in configuration and need assistance.
My callflow: Telco – PRI – GW – H323– CUCM – SIP – ATA187 – Modem
Voicegateway (Version 15.3(2)T) + CUCM (Version= 8.6) + ATA187 (Version= 9.2.3.1)
The modem connection is still not working.
What is still to configure on the voicegateway? modem passthrough?
Regards Michael
ATA 187 Configuration:
Fax Mode= T.38 Fax Relay
Fax Error Correction Mode Override= Off
Maximum Fax Rate= 14000bps
Impedance= 900Ohms complex
Gateway Configuration:
voice service voip
ip address trusted list
ipv4 172.30.50.1
ipv4 172.30.50.2
ipv4 172.30.50.3
ipv4 172.30.50.4
ipv4 172.30.50.5
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h225-notify cid-update
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 600 min 60
dial-peer voice 1 pots
translation-profile incoming INCOMING_PSTN
incoming called-number .
direct-inward-dial
dial-peer voice 30 voip
description OUTGOING_CUCM
destination-pattern [1-9]..
session target ipv4:172.20.60.12
voice-class codec 1
dtmf-relay h245-alphanumeric
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000
no vad

It is possible T38 isn't playing well with the PRI. You could try modem pass-through on the gateway and ATA187 if T38 isn't necessary.
Also, sometimes these commands are needed, but not always, so I would consider whether these fax commands under the dial-peer are necessary:
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000

SIP Interworking difference on UCS or HCS platform ?

SIP Interworking difference on UCS or HCS platform ?
Is there a difference in the SIP interworking off the Cisco CallManagers on a UCS or HCS Platform ?

I suggest do a network capture or enable debug ccsip mesages.
look for conneciion ip address inside sdp field and check that are recheacble.
regards

H323 to SIP calls

Can someone explain how h323 to SIP calls work & vice versa.

The following messages are mapped:
SIP <---> H323
INVITE - SETUP
100 Trying - Call Proc
180 Ringing - Alerting
183 Session Progress - Progress
200 OK (for INVITE) - Connect
BYE - Release Complete
With H323 to SIP CUBE, if fast start occurs on one leg, early offer needs to happen on the other (and vice versa). Most SIP devices these days to early offer (SDP in invite) so you typically need fast start enabled on both directions of the H323 leg for this design.
Check out this link for more information:
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-h323sip_ps5640_TSD_Products_Configuration_Guide_Chapter.html

CME 7.1 with SCCP 7940G phones and SIP connection to a VOIP provider - inbound outbound fails

Here's a quick and dirty diagram of a CME 7.1 configuration. The phone can all call each other but something is not quite right with the SIP provider. The registrar and SIP registration pieces are working but most of the configuration examples that I've seen make me think that the CME router was being used as the edge device to the internet. From my drawing, you can see that is not the case here. My edge device is a Cisco ASA5505 with 9.2.x software running. I might be missing something in the SIP gateway knowledge department. Without diving into the configuration, I'm wondering if SIP messages are failing for calls because of NAT'ing? Trying to do searches has been tricky because I keep running into information that is more about setting up CME for SIP phones or just getting SIP to work between CME and a SIP provider. I have that part working. I'm just a bit unsure about how an SCCP 7940G gets an outbound call or even gets one to come in.
When I dial from my cell phone to the pilot number, there are no rings, it just goes to the VOIP provider's voice mail. When I try to dial out, I get a fast busy.
So, is NAT a consideration? Will the SIP gateway set up a call (forward) via the pre-established SIP connection? Yeah, I do sound like a newb.
If anyone has good information about, let's say, an inbound call and how that traffic flow works.
Thanks!

Have you configured your ASA to either NAT the IP address of the CME router or to do port forwarding for port 5060?

MXP's and SIP Registration Failure Using a FQDN

A majority of our MXP's will not register to our SIP Server using a FQDN. They will register using the SIP Server IP Address. Why do they not like the FQDN? Our C Series, EX, and SX systems do not have this problem.

yes to both questions. some of the MXP's show that they are registered with the FQDN, but that the Active SIP Server Address is the FQDN. I was expecting to see the IP Address of the VCS it is registered to.The majority just won't register with the FQDN, but do with an IP Address.
All of our other non-MXP endpoints register just fine using the FQDN.
The same FQDN is used for Gatekeeper and SIP registration. The MXP's register to the gatekeeper as expected. It is just the SIP Registration.
The majority are running F9.3.3. Some others are running anywhere from F9.02 to F9.3.1, but all software releases are affected the same way.

Adding ICT trunk and SIP trunk into Route group

Hi ,
We need to map ICT and SIP trunk into the same route group ,but the problem here is already same ICT is mapped to another route pattern.
If i try to create new ICT with same remote IP ,it's throwing Add failed because the remote IP is already defined.
Is there anyway we can add ICT with same remote IP and map the ICT and SIP trunk ? or Is there anyway that we can add exisitng ICT into route pattern.
Route pattern is used for this route group is different.CUCM version - 7.X.Please advice.
Regards,
Ramanathan

Thanks Suresh ...
In that case ,I can assign the route group(ICT ,SIP - Top Down) to two different Route pattern.
Both patterns will hit ICT first ,Please correct me if I am wrong.
Ram

H323 and SIP Interworking

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