<br> 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<br><br>voice class codec 1<br> codec preference 1 g711alaw<br> codec preference 2 g711ulaw<br><br><br>voice service voip<br> allow-connections h323 to sip<br> allow-connections sip to h323<br> fax protocol t38 ls-redundancy 2 hs-redundancy 0 fallback none<br> h323<br> session transport udp<br> h245 tunnel disable<br> sip<br><br><br>dial-peer voice 1 voip<br> destination-pattern 26.<br> voice-class codec 1<br> session protocol sipv2<br> session target ipv4:192.168.10.250<br> session transport udp<br><br>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 .<br>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<br>but when i debug the isdn q931 i didn't get the ALERTING message from the PSTN & noticed this message.<br> <br>Unallocated/unassigned number<br>Invalid information element contents<br><br>also i made a sip error debug & i got the following :<br><br>Jan 16 21:39:52.825: //21250/67DAB60FAB28/SIP/Error/sipSPISearchForForkingCodec<br>: Non-conforming codec (g711alaw) in stream 2; the stream will be rejected.<br>SIP: (21250) Attribute ptime, level 1 instance 1 not found.<br>SIP: (21250) Attribute ptime, level 2 instance 1 not found.<br>SIP: (21250) Attribute ptime, level 2 instance 1 not found.<br><br><br><br>*Jan 16 21:43:46.509: //21252/E070770FAB2A/SIP/Call/sipSPICallInfo:<br>The Call Setup Information is:<br>Call Control Block
(CCB) : 0x65F1F2D8<br>State of The Call : STATE_DEAD<br>TCP Sockets Used : NO<br>Calling Number : no_from_info<br>Called Number : 933478244<br>Source IP Address (Sig ): 192.168.10.150<br>Destn SIP Req Addr:Port : 192.168.10.250:5060<br>Destn SIP Resp Addr:Port : 192.168.10.250:5060<br>Destination Name : 192.168.10.250<br><br>*Jan 16 21:43:46.509: //21252/E070770FAB2A/SIP/Call/sipSPIMediaCallInfo:<br>Number of Media Streams: 2<br>Media Stream : 1<br>Negotiated Codec : g711ulaw<br>Negotiated Codec Bytes : 160<br>Negotiated Dtmf-relay : 0<br>Dtmf-relay
Payload : 0<br>Source IP Address (Media): 192.168.10.150<br>Source IP Port (Media): 19596<br>Destn IP Address (Media): 192.168.10.250<br>Destn IP Port (Media): 56476<br>Orig Destn IP Address:Port (Media): 0.0.0.0:0<br><br>*Jan 16 21:43:46.509: //21252/E070770FAB2A/SIP/Call/sipSPIMediaCallInfo:<br>Number of Media Streams: 2<br>Media Stream : 2<br>Negotiated Codec : g711alaw<br>Negotiated Codec Bytes : 160<br>Negotiated Dtmf-relay : 0<br>Dtmf-relay Payload : 0<br>Source IP Address (Media): 192.168.10.150<br>Source IP Port (Media): 0<br>Destn IP Address (Media): 192.168.10.250<br>Destn IP Port (Media): 0<br>Orig Destn IP Address:Port (Media): 0.0.0.0:0<br><br>*Jan
16 21:43:46.509: //21252/E070770FAB2A/SIP/Call/sipSPICallInfo:<br>Disconnect Cause (CC) : 3<br>Disconnect Cause (SIP) : 404<br><br><br><br>can anyone help me in this case , any ideas<br>thanks you all :)<br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><p> 
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