- This topic has 2 replies, 1 voice, and was last updated 10 years, 6 months ago by MikeM to Michael Soo.
16th July 2009 at 17:23 #31893Abhi NavreGuest
Below is an event log for a call coming from an asterisk server which needs to be routed through the quintum gateway. Now the gateway is connected on its first fxs port to a vonage ata which will be placing the calls. but i am encountering this problem of no routes corresponding to TG=0 being available and a disconnect error code 34 is showing up. I need help regarding what this TG=0 is and how do I go about this issue. Below is the event log for the tenor af:
chsip: negotiated media ptime 0
chsip: Setting remote rtp port=10.5
Remote side packet saver version = 2.
sip[0/0]: chsiptcall:doTranslation inc
SipOrigCall : TON=0 and NPI=0.
ocall : the callinfo guid IS 0, 0,
CallInfo : origCalled.digit(64694
24754) callingparty (1011)
sent message to sip: msg=9; ua=1
Routing requested for:
Orig#=6111111111 NPI=0(public) TON=0
Normalized#=16111111111 NPI=1(public) TON=1
Incoming SRC:10.x.x.x CallingParty:1
Route code= selected TG=0
RouteInfo : Ext route requested.
excp |01/01| 2009/07/14|20:57:02:330 |> CH < Radius: Missing radius config uration parameters > CH < RadiusRequest: missing requir ed radius parameter. ch |01/01| 2009/07/14|20:57:02:330 |Ext route req(117): send() failed. 0 match(es) found: Route response : result=0 cause=3 4. use the cause from previous attempt if one available ch |01/01| 2009/07/14|20:57:02:335 |CallInfo: fail event. cause=34 le gno=0 leg=0 sentLeg=0. CallInfo : discTickm(4506861) con nTickm(0) duration (0) . CallInfo: send eventFailed 0. sip[117/0]: osipcall:stackSendRelease ch |01/01| 2009/07/14|20:57:02:340 |sent message to sip: msg=6; ua=1 sip[0/0]: sipTG::orig releaseCall:call Id does not exist ch |01/01| 2009/07/14|20:57:02:385 |sip[0/0]: chsipTG: SIP CallID :3001e60 a0f861d556b64c2b120974724 sip[0/0]: chsipTG: Sip msg (0x4) rcvd at non-exist callId:(3001e60a0f861d556b64c2b120974724)9th December 2011 at 03:18 #31894Michael SooGuest
Is there a solutions to this matter already?13th December 2011 at 21:51 #31895MikeM to Michael SooGuest
This is most likely either a configuration issue or the quintum is not properly registered with the sip server (if using SIP).
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