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SIP Conversion - No longer able to make modem calls

We are working to convert our voice network off of PRI's and on to SIP Trunks.

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Call Flow:

ITSP—SIP—CUBE—H323—CUCM—H323—Branch Router(2911 or 2811)—Analog Device (Cash Advance Machine) on FXS port.

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All locations that have converted to SIP are now unable to connect to the analog toll free number that the analop Cash Advance Machine Dials.  Getting the following error: SIP/2.0 488 Not Acceptable Media.  Debugs are below for CUBE device.  We have the dial peers set at the branch level to send G.711.  Within CUCM we have the region settings at

Factory Default lossy 64 kbps (G.722, G.711) 384

As you cas see in the logs it is still coming in as below and should probably be G.711:

a=rtpmap:4 G723/8000

a=ptime:30

a=rtpmap:18 G729/8000

Can anyone suggest why the codec is being compressed?  I assume this is why the call setup is failing.

Thanks for the help! 

**I have also attached a file for CUCM debug.  Calling Number = 317-596-8401   Called Number =  800-741-3737

Here are some debugs ran:

Feb  3 15:53:55.821 cst: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:918007413737@32.2XX.X.XX:5060 SIP/2.0
Via: SIP/2.0/TCP 172.1.1.1:5060;branch=z9hG4bK593e211460ba5
From: "7652847084" <sip:7652847084@172.1.1.1>;tag=873207~9e0b435f-333c-412b-8c59-6a746bd381e5-57091540
To: <sip:918007413737@32.2XX.X.XX>
Date: Mon, 03 Feb 2014 21:53:55 GMT
Call-ID: abf1a100-2f010ff3-1e66f-b8112ac@
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:172.1.1.1:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 2884739328-0000065536-0000081835-0193008300
Session-Expires:  1800
P-Asserted-Identity: "7652847084" <sip:7652847084@172.1.1.1>
Remote-Party-ID: "7652847084" <sip:7652847084@172.1.1.1>;party=calling;screen=yes;privacy=off
Contact: <sip:7652847084@172.1.1.1:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 275

v=0
o=CiscoSystemsCCM-SIP 873207 1 IN IP4 172.1.1.1
s=SIP Call
c=IN IP4 10.200.130.53
b=TIAS:8000
b=AS:8
t=0 0
m=audio 18874 RTP/AVP 18 4 101
a=rtpmap:4 G723/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Feb  3 15:53:55.821 cst: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TCP 172.1.1.1:5060;branch=z9hG4bK593e211460ba5
From: "7652847084" <sip:7652847084@172.1.1.1>;tag=873207~9e0b435f-333c-412b-8c59-6a746bd381e5-57091540
To: <sip:918007413737@32.2XX.X.Xtag=65597134-9EC
Date: Mon, 03 Feb 2014 21:53:55 GMT
Call-ID: abf1a100-2f010ff3-1e66f-b8112ac@

CSeq: 101 INVITE
Allow-Events: telephone-event
Warning: 304 32.2XX.X.XX "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0


Feb  3 15:53:55.821 cst: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:918007413737@32.2XX.X.XX:5060 SIP/2.0
Via: SIP/2.0/TCP 172.1.1.1:5060;branch=z9hG4bK593e211460ba5
From: "7652847084" <sip:7652847084@172.1.1.1>;tag=873207~9e0b435f-333c-412b-8c59-6a746bd381e5-57091540
To: <sip:918007413737@32.2XX.X.XX>;tag=65597134-9EC
Date: Mon, 03 Feb 2014 21:53:55 GMT
Call-ID: abf1a100-2f010ff3-1e66f-b8112ac@

Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0

Message was edited by: Jake Riecken

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