08-08-2011 05:05 AM - edited 03-16-2019 06:21 AM
Hi all,
I am trying to configure a CUBE using 2811 router as below:
CUCM........<h323>.....CUBE......<sip>......IPTSP
When making outgoing call from IPphone I can hear the other party, but voice from inside is not going through.
Detail:
IPTSP prefered codec: G729
CUCM setup:
MTP: checked
Region: G729 set for the Gateway DP
CUBE setup:
see attach file.
Here is the SIP-UA call info:
SIP UAC CALL INFO
Call 1
SIP Call ID : DC2D2FF4-C0ED11E0-806AD6A3-7DBA46EC@202.84.46.142
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 09604107000
Called Number : 01713037716
Bit Flags : 0x101A0030 0x100000 0x400
CC Call ID : 207
Source IP Address (Sig ): 202.84.46.142
Destn SIP Req Addr:Port : 202.126.120.100:5060
Destn SIP Resp Addr:Port: 202.126.120.100:5060
Destination Name : 202.126.120.100
Number of Media Streams : 2
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 207
Stream Type : voice+dtmf (0)
Negotiated Codec : g729r8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
Media Source IP Addr:Port: 202.84.46.142:18624
Media Dest IP Addr:Port : 202.126.120.100:6232
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Media Stream 2
State of the stream : STREAM_IDLE
Stream Call ID : -1
Stream Type : voice+dtmf (1)
Negotiated Codec : No Codec (0 bytes)
Codec Payload Type : 255 (None)
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: 202.84.46.142:18734
Media Dest IP Addr:Port : 0.0.0.0:0
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Number of SIP User Agent Server(UAS) calls: 0
Debug CCsip error:
*Aug 8 12:07:45.484: //-1/xxxxxxxxxxxx/SIP/Error/sipsdp_add_standard_lines: media_src_address is NULL; c-line is not added
SIP: Attribute ptime, level 1 instance 1 not found.
*Aug 8 12:07:45.540: //207/8048FA320A00/SIP/Error/sipSPISetStreamInfo: Number of active streams is zero (0)!
*Aug 8 12:07:45.568: //207/8048FA320A00/SIP/Error/sipSPISetStreamInfo: Number of active streams is zero (0)!
*Aug 8 12:07:45.572: //207/8048FA320A00/SIP/Error/sipSPISetStreamInfo: Number of active streams is zero (0)!
*Aug 8 12:07:50.020: //207/8048FA320A00/SIP/Error/sact_active_new_message_response: Invalid response for active call, dropping it
Debug CCsip Message: See attached
Please advise what I am doing wrong. Many thanks in advance.
Fazlay
08-10-2011 09:57 AM
Thanks all. I have done some improvement and it is working with some flaw. No need to answer this thred.
Thanks
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