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Honored Contributor
Honored Contributor

Re: extensions registration

mhj wrote:

I didn't see a display >HW4123 on the analyzer.

Using wireshark

-When i dial the extension 4500 on the VoIP phone,

first the VoIP phone sends a SIP invite.

Second the VoIP phone receive a <401 Unauthorized> message

Third when the VoIP phone send another SIP invite, it receives a <not acceptable here> by the VoIP server and it shows and i cans see on the VoIP phone screen <not acceptable here 4500>

Your VoIP server isn't sending the INVITE to the Adtran device. Fix that based on the instructions for the VoIP server. Many different ways to do this, but totally dependent on the setup of the VoIP server. Consult with the server's tech support.

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Honored Contributor
Honored Contributor

Re: extensions registration

mhj wrote:

the <debug voice verbose> command doesn't show anything

That's because the call isn't reaching the Adtran from your VoIP server.

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New Contributor

Re: extensions registration

Hi jayh,

Good news the <debug voice verbose> command worked! The problem was that i was missing the <ip route> command in my config so i added <ip route 0.0.0.0 0.0.0.0 192.168.184.250>.

Before i add the <ip route> command i could call from my Ameritec to my VoIP phone, because i created a trunk in my VoIP server to the Adtran. So what i did is that i deleted this trunk and i put the <ip route> command and now i can see the <debug voice verbose> command trace when i try to call from my VoIP phone to the Ameritec.

here is the trace of the <debug voice verbose> command:

01:41:36.740 TM.T01 01 SipTM_Idle           rcvd SIP call-leg request: INVITE

01:41:36.741 TM.T01 01 SipTM_Idle           call-leg -> Offering

01:41:36.741 TM.T01 01 SipTM_Idle           State change      >> SipTM_Idle->Sip    TM_Trying

01:41:36.741 TM.T01 01 SipTM_Trying         SDP offer is not loopback request

01:41:36.741 TM.T01 01 SipTM_Trying         Processing From for Caller-ID.

01:41:36.742 TM.T01 01 SipTM_Trying         Caller ID Name   = "2211"

01:41:36.742 TM.T01 01 SipTM_Trying         Caller ID Number = "2211"

01:41:36.742 TM.T01 01 SipTM_Trying         info: unable to set redirect number(s) from INVITE

01:41:36.742 TM.T01 01 SipTM_Trying         sent: TA->InboundCall

01:41:36.742 TM.T01 01 Looking up source address for destination 192.168.184.250

01:41:36.742 TM.T01 01 call-leg (0x442a3be8) -> src: 192.168.184.225 : 5060  dst : 192.168.184.250 : 5060

01:41:36.743 TM.T01 01 SipTM_Trying         sent: 100 Trying

01:41:36.743 TA.T01 01 TAIdle               rcvd: inboundCall from TM

01:41:36.743 TA.T01 01 State change      >> TAIdle->TAInboundCall (TAS_Calling)

01:41:36.744 TA.T01 01 Failed - DID translation: no match for 4501, using 4501

01:41:36.744 TA.T01 01 TAIdle               sent: call to SB

01:41:36.744 TM.T01 01 SipTM_Trying         tachg -> TAInboundCall

01:41:36.744 TM.T01 01 SipTM_Trying         State change      >> SipTM_Trying->SipTM_Pending

01:41:36.744 SB.CALL 14 Idle                 Called the call routine with 4501

01:41:36 SB.TGMgr For dialed number 4501, against template $, on TrunkGroup SIP, the score is 500

01:41:36.745 SB.CALL 14 Idle                 No TRUNK accepted dialed number (4501)

01:41:36.745 SB.CALL 14 Idle                 No LOCAL station matched dialed number (4501)

01:41:36.745 SB.CALL 14 Idle                 No routable destination found on call from (2211) to (4501)

01:41:36.745 SB.CALL 14 State change      >> Idle->CallIdlePending

01:41:36.745 TA.T01 01 TAInboundCall        CallResp event accepted

01:41:36.745 TA.T01 01 State change      >> TAInboundCall->TAClearingComplete (TAS_Clearing)

01:41:36.746 TM.T01 01 SipTM_Pending        tachg -> TAClearingComplete

01:41:36.746 TM.T01 01 SipTM_Pending        State change      >> SipTM_Pending-> SipTM_CallFail

01:41:36.747 TM.T01 01 SipTM_CallFail       call-leg -> Disconnected

01:41:36.747 TM.T01 01 SipTM_CallFail       CallLegStateChanged to Disconnected - TM change to closing state.

01:41:36.747 TM.T01 01 SipTM_CallFail       State change      >> SipTM_CallFail- >SipTM_Closing

01:41:36.747 TM.T01 01 SipTM_Closing        sent: TA->Clear

01:41:36.747 TM.T01 01 SipTM_CallFail       sent: 0

01:41:36.748 TM.T01 01 SipTM_Closing        State change      >> SipTM_Closing->   SipTM_Terminated

01:41:36.748 TM.T01 01 SipTM_Terminated     sent: TA->AppearanceOff

01:41:36.748 TM.T01 01 SipTM_Terminated     State change      >> SipTM_Terminated->SipTM_Idle

01:41:36.749 TA.T01 01 TAClearingComplete   rcvd: clear from TM

01:41:36.749 TA.T01 01 TAClearingComplete   rcvd: appearance off from TM

01:41:36.749 TA.T01 01 TAClearingComplete   Clear Local Variables

01:41:36.750 TA.T01 01 State change      >> TAClearingComplete->TAIdle (TAS_Idle )

01:41:36.750 TM.T01 01 SipTM_Idle           tachg -> TAIdle

01:41:36 SB.CallStructObserver 14 Created

01:41:36 SB.CallStructObserver 14 <-> 37012a8d0dc87ad42697fc4312dc5260@192.168.184.250:5060

01:41:36 SB.CallStructObserver 14 Finalized

2000.01.04 01:41:37 SMDR 14         01/04/2000 01:41:36      0.0 0    E  00/00 2 211            2211            00/00 Unknown         4501            0 N

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Honored Contributor
Honored Contributor

Re: extensions registration

mhj wrote:

01:41:36.744 TA.T01 01 TAIdle sent: call to SB

01:41:36.744 TM.T01 01 SipTM_Trying tachg -> TAInboundCall

01:41:36.744 TM.T01 01 SipTM_Trying State change >> SipTM_Trying->SipTM_Pending

01:41:36.744 SB.CALL 14 Idle Called the call routine with 4501

01:41:36 SB.TGMgr For dialed number 4501, against template $, on TrunkGroup SIP, the score is 500

01:41:36.745 SB.CALL 14 Idle No TRUNK accepted dialed number (4501)

01:41:36.745 SB.CALL 14 Idle No LOCAL station matched dialed number (4501)

01:41:36.745 SB.CALL 14 Idle No routable destination found on call from (2211) to (4501)

OK, so it didn't go to trunk T02.

Is the T1 connected and showing as up?

"show interface t1 0/1" and "show voice trunk T02" should give a hint.

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New Contributor

Re: extensions registration

I can make calls from both side. The problem was the VoIP phone, it can receive a call but cannot initiate any call, i had to change it.

Is it possible to give to tag every time slot on a T1 interface to an extension that i register ? For example, i want to make timeslot 1 usable only for the extension 4500, and timeslot 2 only for the extension 4501.

Thanks

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New Contributor

Re: extensions registration

So i would like to personalize the resource selection and not leave it to hunting methods

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Honored Contributor
Honored Contributor

Re: extensions registration

Create a trunk for each channel and a grouped-trunk including that trunk with the appropriate accept statement.

voice trunk T02 type t1-rbs supervision wink role network

  description "PBX RBS"

  resource-selection linear ascending

  did digits-transferred 4

  connect t1 0/1 tdm-group 1

!

voice trunk T03 type t1-rbs supervision wink role network

  description "PBX RBS"

  resource-selection linear ascending

  did digits-transferred 4

  connect t1 0/1 tdm-group 2

!

voice grouped-trunk PBX

  trunk T02

  accept 4500 cost 0

!

voice grouped-trunk PBX

  trunk T03

  accept 4501 cost 0

And so on...

Note: You really don't need resource-selection linear ascending as there's only one voice path per trunk.

And you might want did digits-transferred 0 as there's no need to send digits. Each DS0 is mapped uniquely.

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New Contributor

Re: extensions registration

Thank you very much jayh !

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