OXE sip trunk reply 404 rejected/not found here

[alexeik]

basicaly did this work before? or is it completely new? how is your sipextgw configured? and what are your used ip-adresses (OXE and SBC)?

[morio]

hi sadim,
have changed it and done a reboot ,, but yet the same,

i got a reply from ALE tech support as below,,,
''
For the destination, the REQ URI is used. The host of REQ URI “itc.sa” is not recognized in the OXE
suggestion: please make modifications in the SBC to have the IP@ of the OXE ( 192.168.54.10) instead of the host in the REQ URI of the INVITE''

maybe they are assuming im using a alcatel SBC, but i am not, im using a cisco 1941 and as per my network engineers they say this can not be done by it.

any ideas guys???

[sadim]

If you have a DNS server, you can try configuring it on the OXE SIP part.

[morio]

hi,
done with the dns server as well,, no luck

[alexeik]

hi sadim,
have changed it and done a reboot ,, but yet the same,

i got a reply from ALE tech support as below,,,
''
For the destination, the REQ URI is used. The host of REQ URI “itc.sa” is not recognized in the OXE
suggestion: please make modifications in the SBC to have the IP@ of the OXE ( 192.168.54.10) instead of the host in the REQ URI of the INVITE''

maybe they are assuming im using a alcatel SBC, but i am not, im using a cisco 1941 and as per my network engineers they say this can not be done by it.

any ideas guys???

we use audiocodes sbc, and they are capable of manipultaing these URI. Maybe your SBC is not compatible with an Alcatel OXE

[sadim]

Can you show the sip ext gw configuration?
In the sip ext gw, is the sip remote domain equal to 10.224.7.20 ?

[morio]

this is my local gw for oxe

qReview/Modify: SIP Gatewayqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqk
x x
x Node Number (reserved) : 1 x
x Instance (reserved) : 1 x
x Instance (reserved) : 1 x
x x
x SIP Subnetwork : 9 x
x SIP Trunk Group : 0 x
x IP Address : 192.168.54.10 x
x Machine name - Host : node001 x
x SIP Proxy Port Number : 5060 x
x SIP Subscribe Min Duration : 1800 x
x SIP Subscribe Max Duration : 86400 x
x Session Timer : 1800 x
x Min Session Timer : 900 x
x Session Timer Method + UPDATE x
x DNS local domain name : --------------------------------------- x
x DNS type + DNS A x
x SIP DNS1 IP Address : --------------------------------------- x
x SIP DNS2 IP Address : --------------------------------------- x
x SDP in 18x + True x
x CAC SIP-SIP + False x




external sip gw
Review/Modify: SIP Ext Gatewayqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqk
x x
x Node Number (reserved) : 1 x
x Instance (reserved) : 1 x
x SIP External Gateway ID : 1 x
x x
x Gateway Name : itc x
x SIP Remote domain : 217.170.63.20 x
x PCS IP Address : --------------------------------------- x
x SIP Port Number : 5060 x
x Transport type + UDP x
x Belonging Domain : itc.sa x
x Registration ID : --------------------------------------- x
x Registration ID P_Asserted + False x
x Registration timer : 0 x
x SIP Outbound Proxy : --------------------------------------- x
x Supervision timer : 0 x
x Trunk group number : 1 x
x Pool Number : -1 x
x Outgoing realm : --------------------------------------- x
x Outgoing username : ---------------------------------------
x
x Outgoing Password : -------------------- x
x Confirm : -------------------- x
x x
x Incoming username : --------------------------------------- x
x x
x Incoming Password : -------------------- x
x Confirm : -------------------- x
x x
x RFC 3325 supported by the distant + True x
x DNS type + DNS A x
x SIP DNS1 IP Address : --------------------------------------- x
x SIP DNS2 IP Address : --------------------------------------- x
x SDP in 18x + False x
x Minimal authentication method + SIP None x
x INFO method for remote extension + False x
x To EMS + False x
x SRTP + RTP only x
x Ignore inactive/black hole + False x
x Contact with IP address + False x
x Dynamic Payload type for DTMF : 97
x
x Outbound Calls 100 REL + Supported x
x Incoming Calls 100 REL + Not Requested x
x Gateway type + Standard type x
x Re-Trans No. for REGISTER/OPTIONS : 2 x
x P-Asserted-ID in Calling Number + True x
x Trusted P-Asserted-ID header + True x
x Diversion Info to provide via + History Info x
x Proxy identification on IP address + True x
x Outbound calls only + False x
x SDP relay on Ext. Call Fwd + Default x
x SDP Transparency Override + False x
x RFC 5009 supported / Outbound call + Not Supported x
x Nonce caching activation + NO x
x FAX Procedure Type + T38 to G711 fallback x
x DNS SRV/Call retry on busy server : 0 x
x Unattended Transfer for RSI + NO x
x Redirection functionality + NO x
x Attended Transfer + NO x
x Send BYE on REFER + YES x
x Support Redirection response + NO
Send BYE on REFER + YES x
x Support Redirection response + NO x
x OPTIONS required + YES x
x Support UTF8 characters set + NO x
x Support CSTA User-to-User + NO x
x DDI destination number + ReqURI x
x Video Support Profile + Not Supported x
x UPDATE in Allow header/INVITE + Optional x
x RFC 4904 supported + NO x
x Bulk registration (RFC 6140) + NO x
x RFC3264 m-line + True x
x Sendonly for hold + False x
x In Band DTMF + NO x
x Trusted From header + True x
x Support Re-invite without SDP + True x
x Type of codec negotiation + Single codec G711 x
x
x hope this will useful,,

[sadim]

Can you try to create another Sip Ext Gw equal to the SIP Ext Gw nbr 1, but with "SIP Remote domain : 10.224.7.20"

[haroun]

And when u resolve the DNS/Reverse DNS names u get answer from you DNS server?! I see that there is no DNS server A @ip at the local sip gateway

[morio]

i used a dns as well and it didnt resolve the issue. but now the provider will send us by the ip address but yet i have the issue. ALE tech support advices to remove the ''Route'' in the invite and this must be done in the SBC, but unfortunately im using a router where i cant remove this route...