Can we assign static extensions to each of the 24 timeslots on the t1 trunk side.
I'm not familiar with that exact platform, but if it's similar to the TA900 you can assign a separate voice trunk to each timeslot and configure each with a unique DNIS. Dialing that number will send ringing out that specific channel.
T-1 RBS/CAS connections don't exactly have "registration" in the sense that SIP endpoints do.
Actually i have a telephone simulator (Ameritec PCM/VF call analyzer) plugged into my NetVanta 644 through my t1 connection, and a ethernet cable from my gigabit-ethernet port that goes into a switch connected to a VoIP server and VoIP phone.
I can make calls from my Ameritec to my VoIP phone by dialing the VoIP phone extension. But to call from my VoIP phone to the Ameritec i need an extension and the problem is that the Ameritec doesn't have an extension or a number it is a phone simulator, so i am trying to find a way to give an extension to the t1 trunk, so when the adtran receive a call from a number it puts it on the t1 line.
Try something like this, assuming that the T1 voice trunk is T01. Dialing 4444 should connect to the simulator.
!
voice grouped-trunk T1-test
trunk T01
accept 4444
!
If this doesn't work, paste your configuration.
If you want individual extensions for each channel, you'll need to create 24 voice trunks, each connected to one channel (tdm-group), then build 24 grouped-trunks associated with the voice trunks and each having a unique extension.
The usual way to do this is either with a PRI that can dynamically route the extensions, or with a single RBS trunk and DIDs where the extension digits are sent to the PBX as DTMF. The PBX then routes the calls based on received digits. If you'll be connecting to a channel bank with a unique extension per channel, then you'll need the individual trunks and extensions.
I'm not familiar with your simulator and whether it can detect and display DTMF, or act as a PRI. If it can, then you can play with the various options.
this is my network topology.
I dont have a PBX for the Ameritec. This is why i am trying to find a way to pass the call without a PBX. I am trying to find if it is possible.
you said to build 24 grouped-trunks associated with the voice trunks and each having a unique extension. This might be the solution, how can i give unique extension to the trunks as you said.
thanks
This is my configuration:
644#show run
Building configuration...
!
!
! ADTRAN, Inc. OS version R10.9.6.E
! Boot ROM version A5.01.B2
! Platform: NetVanta 644, part number 1700144G1
! Serial number LBADTN1701AK203
!
!
hostname "644"
enable password nv644
!
clock no-auto-correct-DST
!
ip subnet-zero
ip classless
ip routing
ipv6 unicast-routing
!
!
no auto-config
!
event-history on
no logging forwarding
no logging email
!
no service password-encryption
!
username "admin" password "nv644"
!
!
no ip firewall alg msn
no ip firewall alg mszone
no ip firewall alg h323
!
!
interface gigabit-eth 0/1
ip address 192.168.184.225 255.255.255.128
media-gateway ip primary
no shutdown
!
!
interface gigabit-eth 0/2
no ip address
shutdown
!
!
!
!
interface t1 0/1
timing-domain 1
tdm-group 1 timeslots 1-24 speed 64
no shutdown
!
interface t1 0/2
timing-domain 1
no shutdown
!
interface t1 0/3
timing-domain 1
no shutdown
!
interface t1 0/4
timing-domain 1
no shutdown
!
!
no tftp server
no tftp server overwrite
http server
no http secure-server
no snmp agent
no ip ftp server
no ip scp server
no ip sntp server
!
!
sip
sip udp 5060
no sip tcp
!
!
!
voice feature-mode network
voice forward-mode network
!
!
voice trunk T01 type sip
sip-server primary 192.168.184.250
!
voice trunk T02 type t1-rbs supervision wink role network
connect t1 0/1 tdm-group 1
rtp delay-mode adaptive
!
!
voice grouped-trunk SIP
trunk T01
accept $ cost 0
!
!
voice grouped-trunk T1
trunk T02
accept $ cost 0
!
!
voice grouped-trunk F
2000.01.01 02:48:31 SIP.STACK ERROR MSGBUILDER SIP Pre-Parser Error (UDP) :
!
!
line con 0
no login
!
line telnet 0 4
login
password nv644
no shutdown
line ssh 0 4
login local-userlist
no shutdown
!
!
end
OK, because you have a call analyzer and not a PBX, your device doesn't route calls in the normal way. Try the following in your config:
interface t1 0/1
tdm-group 1 timeslots 1 speed 64
tdm-group 2 timeslots 2 speed 64
tdm-group 3 timeslots 3 speed 64
tdm-group 4 timeslots 4 speed 64
tdm-group 5 timeslots 5 speed 64
tdm-group 6 timeslots 6 speed 64
tdm-group 7 timeslots 7 speed 64
tdm-group 8 timeslots 8 speed 64
tdm-group 9 timeslots 9 speed 64
tdm-group 10 timeslots 10 speed 64
tdm-group 11 timeslots 11 speed 64
tdm-group 12 timeslots 12 speed 64
tdm-group 13 timeslots 13 speed 64
tdm-group 14 timeslots 14 speed 64
tdm-group 15 timeslots 15 speed 64
tdm-group 16 timeslots 16 speed 64
tdm-group 17 timeslots 17 speed 64
tdm-group 18 timeslots 18 speed 64
tdm-group 19 timeslots 19 speed 64
tdm-group 20 timeslots 20 speed 64
tdm-group 21 timeslots 21 speed 64
tdm-group 22 timeslots 22 speed 64
tdm-group 23 timeslots 23 speed 64
tdm-group 24 timeslots 24 speed 64
no shutdown
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
connect t1 0/1 tdm-group 2
connect t1 0/1 tdm-group 3
connect t1 0/1 tdm-group 4
connect t1 0/1 tdm-group 5
connect t1 0/1 tdm-group 6
connect t1 0/1 tdm-group 7
connect t1 0/1 tdm-group 8
connect t1 0/1 tdm-group 9
connect t1 0/1 tdm-group 10
connect t1 0/1 tdm-group 11
connect t1 0/1 tdm-group 12
connect t1 0/1 tdm-group 13
connect t1 0/1 tdm-group 14
connect t1 0/1 tdm-group 15
connect t1 0/1 tdm-group 16
connect t1 0/1 tdm-group 17
connect t1 0/1 tdm-group 18
connect t1 0/1 tdm-group 19
connect t1 0/1 tdm-group 20
connect t1 0/1 tdm-group 21
connect t1 0/1 tdm-group 22
connect t1 0/1 tdm-group 23
connect t1 0/1 tdm-group 24
voice grouped-trunk PBX
trunk T02
accept 4XXX cost 0
Now, if you dial any 4-digit extension between 4000 and 4999 it should show up on your call analyzer. For example, calling 4123 should look like:
>HW4123
You'll need to determine how to tell the AM8a to answer the call. I'm not sure that it's capable of doing so. It's primarily a piece of test equipment to analyze traffic and not really intended to be used as a telephone. I found its operation manual online here: http://dedisol.unsads.com/~squalyl/kurteries/manuals/Other/AMERITEC%20AM8a%20PCM%20VF%20Instruction....
You might want to get a channel bank or better yet, a PBX with a T1 PRI/RBS interface for your lab.
I tried but the call didn't go through. It may be the fact that there is no PBX.
Did you see a display on the analyzer similar to >HW4123 ?
I don't think the call will "go through" but the simulator should display the attempt.
What is the output from "debug voice verbose" when you place a call to the simulator? Does it route to T02?
I didn't see a display >HW4123 on the analyzer.
Using wireshark
IP @:
VoIP phone: 192.168.184.246
VoIP server: 192.168.184.250
Adtran interface: 192.168.184.225
I had this result:
-the extension 4500 registered
-When an <Option request> is sent from the VoIP server to the Adtran it reply with the status <501 not Implemented>
-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>
Here is my configuration:
interface t1 0/1
tdm-group 1 timeslots 1 speed 64
tdm-group 2 timeslots 2 speed 64
tdm-group 3 timeslots 3 speed 64
tdm-group 4 timeslots 4 speed 64
tdm-group 5 timeslots 5 speed 64
tdm-group 6 timeslots 6 speed 64
tdm-group 7 timeslots 7 speed 64
tdm-group 8 timeslots 8 speed 64
tdm-group 9 timeslots 9 speed 64
tdm-group 10 timeslots 10 speed 64
tdm-group 11 timeslots 11 speed 64
tdm-group 12 timeslots 12 speed 64
tdm-group 13 timeslots 13 speed 64
tdm-group 14 timeslots 14 speed 64
tdm-group 15 timeslots 15 speed 64
tdm-group 16 timeslots 16 speed 64
tdm-group 17 timeslots 17 speed 64
tdm-group 18 timeslots 18 speed 64
tdm-group 19 timeslots 19 speed 64
tdm-group 20 timeslots 20 speed 64
tdm-group 21 timeslots 21 speed 64
tdm-group 22 timeslots 22 speed 64
tdm-group 23 timeslots 23 speed 64
tdm-group 24 timeslots 24 speed 64
no shutdown
voice trunk T01 type sip
sip-server primary 192.168.184.250
register range 4500 4501
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
connect t1 0/1 tdm-group 2
connect t1 0/1 tdm-group 3
connect t1 0/1 tdm-group 4
connect t1 0/1 tdm-group 5
connect t1 0/1 tdm-group 6
connect t1 0/1 tdm-group 7
connect t1 0/1 tdm-group 8
connect t1 0/1 tdm-group 9
connect t1 0/1 tdm-group 10
connect t1 0/1 tdm-group 11
connect t1 0/1 tdm-group 12
connect t1 0/1 tdm-group 13
connect t1 0/1 tdm-group 14
connect t1 0/1 tdm-group 15
connect t1 0/1 tdm-group 16
connect t1 0/1 tdm-group 17
connect t1 0/1 tdm-group 18
connect t1 0/1 tdm-group 19
connect t1 0/1 tdm-group 20
connect t1 0/1 tdm-group 21
connect t1 0/1 tdm-group 22
connect t1 0/1 tdm-group 23
connect t1 0/1 tdm-group 24
voice grouped-trunk PBX
trunk T02
accept 4500 cost 0
voice grouped-trunk SIP
trunk T01
accept $
the <debug voice verbose> command doesn't show anything
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.
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.
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
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.
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
So i would like to personalize the resource selection and not leave it to hunting methods
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.
Thank you very much jayh !