Hi every one , I’m testing a dmg1008lsw ,for a small scenario of UM and OCS , i have set up related to OCS and UM , everything ok , thumbs UP¡
But with the telephony integration I’m having some troubles. My scenario is really small, and the media gateway is as follows¡ PSTN->Gateway->Mediation
Im able to make outbound calls, pstn phones rings two times then the call hangs on, if you can answer on before the 2 rings , you can establish the call. In cell phone calls , the call rings only one time and you have to answer in that ring to make call establish after one ring the call hangs on.
i reach this point by making some changes in tone detection, because before i modified the tone detection , i wasn’t able to make no calls.
This trouble is for outbound calls, for inbound calls i have a different situation, i configured the inbound calls to pass-through directly to the auto attendant of exchange, this auto attendant is access by a simple OCS extension, and the call have to go through mediation Server and no traffic is seen in this server.
The inbound call just keep ringing and ringing , and the gateway never answer .
I’m no to familiarized with deep gateway knowledge, I’m most of all mcse and ocs mcts of OCS.....in other words im stocked with this. Any Help advice would be really appreciate
I attached config file and inbound and outbound call log
Tanx in advanced
The IB call (trace_inboundcalllog.log) is failing because you do not have a INBOUND TDM route setup that matches the caller id information for the call.
Your only IB rule matches on redirect number being: +2000
Try changing this to match on any (*)
Hope this helps
Tanx Vince, I was missing the term of redirect at the inbound tdm request matching , the right place to redirect the call to an specific number in Voip side , is at the outbound route at tdm inbound rule , finally I can get traffic trough my sip server (mediation) , but still having troubles.
I decide to test the inbound calls to see if finally my problem was resolved, configured the gateway to dial to my ocs tel URI then I dialed to my inbound analog trunk line, MOC finally gets ringing but once I answer, another call enters to my moc and another and another and so on, every call keeps establish and if I answer another the last one get resume, as a known functionality of ocs.
Seems like the gateway never knows when the MOC answer the call and keep sending sip invitation to the mediation sever.Additional to this I saw a SIP/2.0 422 Session timer is too small message in the log, don’t know if it’s relevant or not.
I attached a log for that call
About outbound call log, any comments?
From the outbound trace (trace_CellPhoneOutboundCall.log), everything appears to be working as expected. The DMG recieves the INVITE, finds a OB route, goes off-hook, detects dialtone and then proceeds to dial. But from the trace, 5 seconds after the dial completes the DMG hears another dialtone which trips the disconnect event - I dont know why this dialtone is coming after the dialing.... can you provide any ideas?
115:59.564 [VoIP ] Prot ---->INVITE sip:+email@example.com;user=phone SIP/2.0
115:59.628 [RouteTable] Code rtGetOutboundRoute() outbound call info: [+2005,]->[+0448114849237,]->[,] [Rsn=1]
115:59.674 [Tel-2 ] Event Offhook
116:00.958 [Tel-2 ] Event dialtone On
116:00.960 [Tel-2 ] Event KEY_PRS 0
116:01.160 [Tel-2 ] Event KEY_PRS 4
116:01.360 [Tel-2 ] Event KEY_PRS 4
116:01.560 [Tel-2 ] Event KEY_PRS 8
116:01.760 [Tel-2 ] Event KEY_PRS 1
116:01.960 [Tel-2 ] Event KEY_PRS 1
116:02.160 [Tel-2 ] Event KEY_PRS 4
116:02.360 [Tel-2 ] Event KEY_PRS 8
116:02.560 [Tel-2 ] Event KEY_PRS 4
116:02.760 [Tel-2 ] Event KEY_PRS 9
116:02.960 [Tel-2 ] Event KEY_PRS 2
116:03.160 [Tel-2 ] Event KEY_PRS 3
116:03.360 [Tel-2 ] Event KEY_PRS 7
116:03.560 [Tel-2 ] Event dialtone Off
116:08.938 [Tel-2 ] Event dialtone On
The analog lines i attached to the dmg1008 are also dsl lines. The service is give it by Telmex , i also use a noise filter to clear the line, in theroy if i can dial from a normal phone , i must be able to do the same with the dialogic.
Yes - if you can dial from a normal phone the same way the DMG is dialing (i.e. no additional access codes) then it should work with the DMG. Are there any other tones that may be interpreted as dialtone?
Nop , there is no other tones , the lines are pure pstn lines. The dialtone i use for outbound call is learned tone , dont know if i used the tool correctly.
Could it be that somehow the progressing tone that the gateway is getting (the ringback form the far side) could be tagged in the gateway config file as a dial-tone?
Sorry for the lateness to post back, updating , I’m able now to make inbound calls and redirect them to the UM autoatendant ,I make some changes in the config in ringing time out , and get solved the incoming call issue.
Outgoing calls are still in same behavior, in pstn cell phones calls it only ring once and connection is end, in home pstn lines or business pstn lines have several behaviors, in some company’s lines it rings twice and in other company lines it can’t rings indefinitely.
I made a POC for a motor company 2 months ago , with a DMG 2000 series , dmg ->pbx->pstn and something similar happens , the solution for this was give it by dialogic , the partner who back up the POC scale to dialogic and they send a modified version of files to make disappear this kind of errors.
Don’t have a progressing tone tagged as a dial tone, there’s some documentation of how to use the tool of learning tones, maybe I can make the gateway to learn the correct tone of ring back.
later I post again a new log and config file for a review
Tanx to all again¡¡