06-25-2012 01:21 PM - edited 03-16-2019 11:50 AM
I have a SIP trunk from a TSP terminated on the CUBE. - c2901 gateway connected via H323 to a CUCM 8.5 cluster. The signalling looks OK but I have no voice. The provider seems to support only G.729br8 codec and my phones (c6901 and c6921) seem to talk G.729r8 only so I need to do some transcoding. I'm in the UK btw.Now, I can do it on the CUBE and utilize dspfarm but that is I think only possible with SCCP. How do I register the gateway in CUCM, if it's already there as the H323 gateway?
Is there any way to do the transcoding on the CUCM? Any general advise would be much appreciated.
Cheers
Solved! Go to Solution.
06-27-2012 01:58 AM
Hi,
First of I didnt miss the session target command, I omitted because you dont need. This is an incoming called number dial-peer. You dont not have a destination-pattern on this dial-peer so you dont need session target.
Secondly, I dont see the benefit of you keeping h323. I helped resolved a problem similar to this on this forum by changing to SIP. So i strongly suggest you consider this..
Thirdly, I cdan see a disconnect cause of 47
Sent:
CANCEL sip:01413330661@146.191.243.4:6280 SIP/2.0
Via: SIP/2.0/UDP 146.191.201.41:5060;branch=z9hG4bK217CC
From: "Michelle Walters" <01387345811>;tag=40B9354-94401387345811>
To: <01413330661>01413330661>
Date: Wed, 27 Jun 2012 08:23:56 GMT
Call-ID: 45C8BC2C-BF6811E1-80BAA4DC-D76223A2@146.191.201.41
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1340785438
Reason: Q.850;cause=47
This is usually points to a codec issue.
We need to know what is happening on the call leg to cucm. Because you are using h323, you need to send debug voip ccapi inout.
I honestly advise changing to sip.
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-27-2012 02:14 AM
OK, if I go SIP from cucm to CUBE, do I add the CUBE in cucm as gateway or as trunk? I'll give it a try. Thanks again. The reason why I started off with h323 is we have other gateways configured like this:
cucm -->h323 -->gateway --> PSTN
I understand it makes sense to keep it SIP all way through.
06-27-2012 02:17 AM
Yes you need to add a sip trunk on cucm as I explained in the other post. Thats all. Then configure your sip binding on your gig0/1 interface as I also explained.
Once you have done this, do a test call and send the debug ccsip messages. With this we will see the interactuion between cucm and cube on the debug.
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-27-2012 02:39 AM
06-27-2012 03:05 AM
Ok,
You still have not configured the dial-peer to CUCM to use SIP. But here is what I see..
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 146.191.243.4:6280;branch=z9hG4bK7kljtp2088b0cmgv34l0.1
From:
To: <5811>;tag=44F7878-C235811>
Date: Wed, 27 Jun 2012 09:38:06 GMT
Call-ID: ODE1Y2E4ZTlhZDg1NjM0MmRhZDE4YmNhNTRiMjhkN2M.
CSeq: 1 INVITE
Require: 100rel
RSeq: 650
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "Michelle Walters" <5811>;party=called;screen=no;privacy=off5811>
Contact: <5811>5811>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 5886 9981 IN IP4 146.191.201.41
s=SIP Call
c=IN IP4 146.191.201.41
t=0 0
m=audio 20748 RTP/AVP 18 96
c=IN IP4 146.191.201.41
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
You are sending G729br8 to your provider but they do not support annexb because in their invite we do not see annexb.
INVITE sip:5811@UniWestScot:5060 SIP/2.0
Via: SIP/2.0/UDP 146.191.243.4:6280;branch=z9hG4bK7kljtp2088b0cmgv34l0.1
Max-Forwards: 68
Contact:
To: <5811>5811>
From:
Call-ID: ODE1Y2E4ZTlhZDg1NjM0MmRhZDE4YmNhNTRiMjhkN2M.
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, REFER, PRACK
Content-Type: application/sdp
Supported: 100rel
P-Asserted-Identity:
Content-Length: 278
v=0
o=Redwood_INX 347015 347015 IN IP4 146.191.243.4
s=Redwood Media Server
c=IN IP4 146.191.243.4
t=0 0
m=audio 50002 RTP/AVP 8 18 96
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000
a=sendrecv
a=silenceSupp:off
a=ecan:on
a=fmtp:96 0-15
So either configure your cube to transcode from g729br8 to g729r8 or send your provider g729r8
Here is the document to use to setup xcoding on your cube..
Ensure you add g729br8 in your list of codecs
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-27-2012 03:54 AM
Hi there,
I've replaced the codec g729br8 with g729r8 - it didn't like it at all - our cube SENT SIP/2.0 488 Not Acceptable Media.
Is it possible the provider doesn't support either of the 2 codecs? I'm having difficulties to get a list from them.
Cheers
06-27-2012 04:04 AM
What do you mean by
You still have not configured the dial-peer to CUCM to use SIP
Is it replacing voice-class h323 with session protocol sipv2 in the CUCM dial peers 5001, 5002, 5555?
06-27-2012 04:26 AM
configure
dial-peer voice 5002 voip
session protocol sipv2
do the same for all dial-peers to cucm.
do a test call and send "debug ccsip all"
remove the voice class h323 command..
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-27-2012 05:01 AM
06-27-2012 05:41 AM
Hi
I got the outbound calls working.
The pronblem is a codec mismatch.
I added
g729 annexb-all
on the CUBE. However, that doesn't fix the inbound calls.
06-27-2012 06:48 AM
Hi, I have looked at the trace again and I can see again the codec issue.
Here is a summary of trace analysis..
+++CUBE sends SDP to ITSP with g729br8+++
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 146.191.243.4:6280;branch=z9hG4bKg8plvd1008sg4n86i2p0.1
From:
To: <5811>;tag=4D1597C-40E5811>
Date: Wed, 27 Jun 2012 11:59:58 GMT
Call-ID: NmRhNmJlZGJkNjZhMmNiYTk1MmFmMjdlOGU0MzU1NDk.
CSeq: 1 INVITE
Require: 100rel
RSeq: 5815
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <5811>;party=called;screen=yes;privacy=off5811>
Contact: <5811>5811>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 260
v=0
o=CiscoSystemsSIP-GW-UserAgent 3479 16 IN IP4 146.191.201.41
s=SIP Call
c=IN IP4 146.191.201.41
t=0 0
m=audio 26048 RTP/AVP 18 96
c=IN IP4 146.191.201.41
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
++++CUBE receives 200 ok from ITSP with G729br8+++So ITSP supports G729br8
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 146.191.201.41:5060;branch=z9hG4bK35E20
From: <5811>;tag=4D1597C-40E5811>
To:
Call-ID: NmRhNmJlZGJkNjZhMmNiYTk1MmFmMjdlOGU0MzU1NDk.
CSeq: 101 INVITE
Timestamp: 1340798408
Contact:
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, REFER, PRACK
Content-Type: application/sdp
Content-Length: 258
v=0
o=Redwood_INX 432119 432238 IN IP4 146.191.243.4
s=Redwood Media Server
c=IN IP4 146.191.243.4
t=0 0
m=audio 50038 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=sendrecv
a=ptime:20
++++CUBE receives 200 ok from cucm with only g729r8 (no annex-b)+++
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 146.191.201.41:5060;branch=z9hG4bK3021ED
From:
To: <5811>;tag=107724~58ef7c6e-6d55-43ea-a324-3eb4beb02da5-752656005811>
Date: Wed, 27 Jun 2012 12:00:12 GMT
Call-ID: 72E7D3B6-BF8611E1-81BEA4DC-D76223A2@146.191.201.41
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence, kpml
Supported: replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uas
Require: timer
P-Asserted-Identity: "Michelle Walters" <5811>5811>
Remote-Party-ID: "Michelle Walters" <5811>;party=called;screen=yes;privacy=off5811>
Contact: <5811>5811>
Content-Type: application/sdp
Content-Length: 182
v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 146.191.2.102
s=SIP Call
c=IN IP4 146.191.199.29
t=0 0
m=audio 16438 RTP/AVP 18
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
+++CUBE rejects call and sends a BYE to CUCM with cause code of 65++++++++++++
Sent:
BYE sip:5811@146.191.2.102:5060 SIP/2.0
Via: SIP/2.0/UDP 146.191.201.41:5060;branch=z9hG4bK32180E
From:
To: <5811>;tag=107724~58ef7c6e-6d55-43ea-a324-3eb4beb02da5-752656005811>
Date: Wed, 27 Jun 2012 11:59:57 GMT
Call-ID: 72E7D3B6-BF8611E1-81BEA4DC-D76223A2@146.191.201.41
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1340798404
CSeq: 102 BYE
Reason: Q.850;cause=65
Now cause code of 65 is saying that CUBE does not support the codec call manager is trying to use.
Can you please configure your voice class codec to include g729r8 (I am not sure if the voice class codec you are using on dial-pee 5001, 500..is voice class codec 1 or 2. use the correct one.
eg
voice class codec 1 or voice class codec 2 (whichever o ne you are using on the dp to cucm
codec preference 1 g729r8
codec preference 2 g729br8
Please test again and send debug ccsip messages. Please attach debug in a text file dontg post here..cause the thread is getting too long
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-27-2012 07:07 AM
Hi,
Thanks a lot for your help.
Stupid but I am unable to find a way how to atach a file to this discussion!
Anyway - the command
g729 annexb-all
did this:
- outbound calls work
- inbound calls still no audio, but codec g729r8 is now accepted by the ITSP.
Here is the inbound trace:
06-27-2012 07:38 AM
Ok, Your codec look okay now as I can see that g729r8 is used all the way from CUCM to your ITSP.
I can also see that your cube has 2 IP addresses..
Can you do a "sh voip rtp connection" when you do an inbound call. From your trace I can see the the ff IPs
10.0.48.98..what is this ip?
10.0.16.21..what is this IP?
146.191.243.4---------ITSP IP
146.191.201.41--------CUBE IP
146.191.251.40---------CUCM IP
146.191.199.29-----IP Phone IP
Your RTP connection should look like this..
146.191.201.41------------->146.191.199.29
146.191.201.41-------------->146.191.243.4
Can the CUBE reach the IP of the Phone 146.191.199.29? Can you ping this IP from the CUBE?
Can you please attach the updated config onf your CUBE. To attach click on the advanced editor on the right corner of the writing window. Then at the bottom you will see where to attach a file
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-27-2012 07:44 AM
Hi, yes the RTP connections are
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 267 268 25010 50026 146.191.201.41 146.191.243.4
2 268 267 31480 16496 146.191.201.41 146.191.199.29
Found 2 active RTP connections
The 2 private IP addresses are coming from the ITSP. Our CUBE has 1 IP
hcpgw01#sh ip int brief
Interface IP-Address OK? Method Status Protocol
Embedded-Service-Engine0/0 unassigned YES NVRAM administratively down down
GigabitEthernet0/0 unassigned YES NVRAM administratively down down
GigabitEthernet0/1 146.191.201.41 YES NVRAM up up
06-27-2012 07:49 AM
Can the CUBE reach the IP of the Phone 146.191.199.29? Can you ping this IP from the CUBE?
Can you please attach the updated config onf your CUBE. To attach click on the advanced editor on the right corner of the writing window. Then at the bottom you will see where to attach a file
Please rate useful posts
"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
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