[cisco-voip] Right Fax Problem

[Moataz Mamdouh]

  I have a problem in rightfax integeration with the call manager 4.2(3) , the right fax is configured to use SIP with the router , so i configured a dial peer to use SIP with a seesion target the ip address of the right fax

voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw


voice service voip
 allow-connections h323 to sip
 allow-connections sip to h323
 fax protocol t38 ls-redundancy 2 hs-redundancy 0 fallback none
 h323
  session transport udp
  h245 tunnel disable
 sip


dial-peer voice 1 voip
 destination-pattern 26.
 voice-class codec 1
 session protocol sipv2
 session target ipv4:192.168.10.250
 session transport udp

the voice gateway was configured to use H.323 with the call manager , all the ip phones are able to send and receive calls based on inbound & oubound dial peers on the H.323 gateway but the right fax is not able to send calls to the PSTN on the E1 Pri .
while debugging the voip dialpeer inout on the gateway i can see the outbound dial peer matching which point to the pots dial peer to the PSTN
but when i debug the isdn q931 i didn't get the ALERTING message from the PSTN & noticed this message.
 
Unallocated/unassigned number
Invalid information element contents

also i made a sip error debug & i got the following :

Jan 16 21:39:52.825: //21250/67DAB60FAB28/SIP/Error/sipSPISearchForForkingCodec
: Non-conforming codec (g711alaw) in stream 2; the stream will be rejected.
SIP: (21250) Attribute ptime, level 1 instance 1 not found.
SIP: (21250) Attribute ptime, level 2 instance 1 not found.
SIP: (21250) Attribute ptime, level 2 instance 1 not found.



*Jan 16 21:43:46.509: //21252/E070770FAB2A/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x65F1F2D8
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : no_from_info
Called Number            : 933478244
Source IP Address (Sig  ): 192.168.10.150
Destn SIP Req Addr:Port  : 192.168.10.250:5060
Destn SIP Resp Addr:Port : 192.168.10.250:5060
Destination Name         : 192.168.10.250

*Jan 16 21:43:46.509: //21252/E070770FAB2A/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 2
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0
Source IP Address (Media): 192.168.10.150
Source IP Port    (Media): 19596
Destn  IP Address (Media): 192.168.10.250
Destn  IP Port    (Media): 56476
Orig Destn IP Address:Port (Media): 0.0.0.0:0

*Jan 16 21:43:46.509: //21252/E070770FAB2A/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 2
Media Stream             : 2
Negotiated Codec         : g711alaw
Negotiated Codec Bytes   : 160
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0
Source IP Address (Media): 192.168.10.150
Source IP Port    (Media): 0
Destn  IP Address (Media): 192.168.10.250
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): 0.0.0.0:0

*Jan 16 21:43:46.509: //21252/E070770FAB2A/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 3
Disconnect Cause (SIP)   : 404



can anyone help me in this case , any ideas
thanks you all :)



















       
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