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2811 Cube [h323-sip] one way voice

fazlayrabby
Level 1
Level 1

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

1 Reply 1

fazlayrabby
Level 1
Level 1

Thanks all. I have done some improvement and it is working with some flaw. No need to answer this thred.

Thanks