I need to configure a 904 to keep calls from one FXS port to another FXS port on the same system internal. AOS 10.7.0
Sample Config:
Trunk t02 type SIP
!Working Sip Config
voice forward-mode local
Voice User 555-1234
connect FXS 0/1
Voice User 555-9876
connect FXS 0/2
voice grouped-trunk SIP
trunk t02
accept $ cost 0
Scenario:
555-1234 goes off hook and dials 555-9876 currently call is sent out SIP Trunk hits upstream provider and then call comes back down stream.
Anyone have any ideas on how to do the call routing to keep the call local if the to number is local to the system?
Desired:
555-1234 goes off hook and dials 555-9876 call never goes out SIP trunk and stays on the local 904
Configuring a specific reject for the local fax numbers on the SIP grouped-trunk still won't cause a problem.
That will prevent calls to those specific numbers from leaving out the SIP trunk. It won't affect incoming calls to those numbers from the SIP trunk.
Seeing as those phone numbers are local FXS users, there is no reason to send calls from the Adtran out the SIP trunk for them. As you've discovered, they'll just hairpin back anyway. So reject them specifically on the SIP trunk and they'll stay local.
This should stay local.
Are both the stations 7-digit and the local calling pattern is also 7-digit?
Did you try it without the hyphen? "voice user 5551234" instead of "voice user 555-1234"?
You might need an alias to sort out the dial plan, for example if the SIP trunk sends 10-digits and you want a 7-digit (or 4-digit extension) local connection. For example:
voice user 3125551234
alias 5551234
alias 1234
connect fxs 0/1
...etc.
Try "debug voice verbose" and make a local call, paste the result as well as the configuration sanitized of passwords.
jayh,
The system would not accept the alias command, i am guessing because the alias and the number where an exact match already I was able to set the DID. In the actual config there is no hyphen I had put that in the sample for readability. In this case all the calls are 10 digit dials. Working on capturing the debug now will post once I have it. Can you or some else verify if it takes a reboot for voice forward-mode local to go into affect. I did change it as originally it was network but changed it to local for this issue.
John Wable
voice forward-mode local has to do with how call-forwarding is handled, not call routing. This shouldn't affect the scenario you describe.
I'm somewhat confused as your sample config showed 7-digits as the voice user and above you state "In this case all the calls are 10 digit dials."
For a call to be handled locally the following three things have to match:
jayh,
Thanks for the reply. I just put in a sample config not the actual config in example thats why it shows 7 digits but here are the excerpts from the actual config based on the three areas you mention with scrubbed NPA-NXX and passwords, in this specific example there is a fax machine at NXXNXX2115 dialing NXXNXX2120 the call goes out the SIP trunk hits the SIP Provider switch is then sent back down the SIP trunk back to the other voice user this is evidenced by two entries in the Call Quality Status with the same From and To number in both which is how I first noticed the calls where not staying local on the Adtran. If it where a voice user misconfiguration I would not expect to see the call come back and connect. The end user was gone for the day so I was not able to capture the debug yet, I will work on getting that tomorrow. I have not seen this problem previously but this is the first time I am doing FXS ports on 10.7 code, all previous Analog deployments had beed on 4.X code.:
voice dial-plan 1 local NXX-NXX-XXXX
voice dial-plan 2 long-distance 1-NXX-NXX-XXXX
voice dial-plan 3 toll-free 1-8XX-NXX-XXXX
voice dial-plan 4 international 011-$
voice codec-list fax
default
codec g711ulaw
voice user NXXNXX2115
connect fxs 0/3
no call-waiting
did "NXXNXX2115"
modem-passthrough
no echo-cancellation
rtp packet-delay fax 150
codec-list fax
voice user NXXNXX2120
connect fxs 0/4
no call-waiting
did "NXXNXX2120"
modem-passthrough
rtp packet-delay fax 150
codec-list fax
This should just work, but I'm curious as to why you have did "NXXNXX2120" in the user configuration. This shouldn't be needed unless you're passing digits to the extension and it may be causing the failure. I would no that out and retry. Debug voice verbose will be helpful.
I wouldn't think it necessary, but if all else fails try adding the actual extension numbers to the dial plan as extensions.
voice dial-plan 5 extensions NXX-NXX-2115
voice dial-plan 6 extensions NXX-NXX-2120
When entering these use the actual digits, not NXX-NXX.
jayh,
Originally the did statements where not there. When I tried to add the alias command as you recommended trying the system would not take the command, so I tried did in place of alias and it took the command. I will try the extensions in the dial plan that does make sense that way the switchboard knows those numbers are local to the Adtran itself.
OK, the alias is needed if the phone needs to ring with two or more different dial patterns, for example if your area is set up for 7-digit local calls and the SIP provider sends you 10-digits to ring the same FXS port you would use the alias of 7-digits to handle the locally dialed calls. Same if you have 3- or 4-digit extensions you want to keep local.
Sometimes you want to have the call hairpin through the provider. For instance if you have a hosted PBX with centralized voice mail offsite you wouldn't be able to leave a message if it stayed local. SABR will handle that case.
Debug voice verbose should give us a better handle on it.
jayh,
Added the extensions to the dial plan using the full actual numbers. Problem is still occurring. Got a copy of the de voice verb got to scrub it. I am seeing the SSwitchboard is matching on NXX-NXX-XXXX rule on sip trunk group and not going any further. I will get it scrubbed and posted shortly.
John Wable
jwable wrote:
Added the extensions to the dial plan using the full actual numbers. Problem is still occurring. Got a copy of the de voice verb got to scrub it but what I did see is the call is connecting to the SIP trunk as soon as it goes of hook before it even starts collecting digits. I will get it scrubbed and posted shortly.
Really? What does the destination string in the SIP INVITE look like?
What does the grouped-trunk that includes the SIP trunk look like? If you just go off-hook and don't dial anything does the dial tone go away after three seconds or so? From your description it sounds almost like there's a hotline auto-ringdown programmed or the SIP trunk is accepting an empty string.
How about posting the config scrubbed of passwords as well as the debug voice verbose. Let us know for the debug what station attempted to dial what string.
Debug Voice Verbose
11:41:57.954 PM.0:2 Idle Processed OFFHOOK
11:41:57.955 PM.0:2 State change >> Idle->Requesting Dialtone
11:41:57.956 SA.5558542245 Ca:0 Idle rcvd: AcctPhoneMgr_appearance(ON) from PM
11:41:57.956 SA.5558542245 Ca:0 Idle State change >> Idle->DigitGathering (CAS_ReqDigits)
11:41:57.957 SA.5558542245 Ca:0 DigitGathering sent: AcctPhoneMgr_cachg(CAS_ReqDigits) to PM
11:41:57.957 PM.0:2 Requesting Dialtone CACHG:ReqDigits on primary CA
11:41:57.958 PM.0:2 State change >> Requesting Dialtone->SendingDigits
11:41:57.961 TONESERVICES.EVENTS fxs 0/2 - empty - Tone Detection: Request resource
11:41:57.961 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: DSP channel allocated for the resource
11:41:57.962 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: constructed
11:41:57.963 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: starting
11:41:57.963 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: TDM map
11:41:58.462 TONESERVICES.EVENTS fxs 0/2 - empty - DialTone Generation: Request resource
11:41:58.463 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: DSP channel allocated for the resource
11:41:58.463 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: constructed
11:41:58.464 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: starting
11:41:58.465 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: TDM map
11:42:00.628 PM.0:2 SendingDigits Digit 7 processed
11:42:00.629 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (5) event
11:42:00.630 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: stopping
11:42:00.630 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: TDM unmap
11:42:00.631 RTP.CHANNEL fxs 0/2 - dsp 0/1.1 - DialTone Generation: releasing RTP resource
11:42:00.632 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: release
11:42:00.632 SA.5558542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(5) from PM
11:42:00.633 SA.5558542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:00.837 PM.0:2 SendingDigits Digit 0 processed
11:42:00.839 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (5) event
11:42:00.839 SA.5558542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(5) from PM
11:42:00.840 SA.5558542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:01.027 PM.0:2 SendingDigits Digit 6 processed
11:42:01.028 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (5) event
11:42:01.029 SA.5558542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(5) from PM
11:42:01.029 SA.5558542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:01.227 PM.0:2 SendingDigits Digit 3 processed
11:42:01.228 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (3) event
11:42:01.229 SA.5558542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(3) from PM
11:42:01.229 SA.5558542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:01.437 PM.0:2 SendingDigits Digit 9 processed
11:42:01.439 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (9) event
11:42:01.439 SA.5558542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(9) from PM
11:42:01.440 SA.5558542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:01.637 PM.0:2 SendingDigits Digit 6 processed
11:42:01.639 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (6) event
11:42:01.639 SA.5558542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(6) from PM
11:42:01.640 SA.5558542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:01.837 PM.0:2 SendingDigits Digit 2 processed
11:42:01.839 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (2) event
11:42:01.839 SA.5558542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(2) from PM
11:42:01.840 SA.5558542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:02.047 PM.0:2 SendingDigits Digit 1 processed
11:42:02.048 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (1) event
11:42:02.049 SA.5558542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(1) from PM
11:42:02.050 SA.5558542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:02.248 PM.0:2 SendingDigits Digit 2 processed
11:42:02.248 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (2) event
11:42:02.249 SA.5558542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(2) from PM
11:42:02.250 SA.5558542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:02.448 PM.0:2 SendingDigits Digit 0 processed
11:42:02.448 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (0) event
11:42:02.449 SA.5558542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(0) from PM
11:42:02.450 SA.5558542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:02.450 SA.5558542245 Ca:0 DigitGathering State change >> DigitGathering->DigitGathering (CAS_Active)
11:42:02.451 SA.5558542245 Ca:0 DigitGathering sent: AcctPhoneMgr_cachg(CAS_Active) to PM
11:42:02.452 PM.0:2 State change >> SendingDigits->Call Pending
11:42:02.454 SA.5558542245 Ca:0 DigitGathering sent: call to SB
11:42:02.454 SA.5558542245 Ca:0 DigitGathering State change >> DigitGathering->CallPending (CAS_Active)
11:42:02.455 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: stopping
11:42:02.456 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: TDM unmap
11:42:02.456 RTP.CHANNEL fxs 0/2 - dsp 0/1.1 - Tone Detection: releasing RTP resource
11:42:02.457 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: release
11:42:02.458 SA.5558542245 rcvd: AcctPhoneMgr_billingCode from PM
11:42:02.459 SB.CALL 11025 Idle Called the call routine with 5553962120
11:42:02 SB.TGMgr For dialed number 5553962120, against template NXX-NXX-XXXX, on TrunkGroup SIP, the score is 1000
11:42:02.460 SB.CCM isMappable:
11:42:02.461 SB.CCM : Call Struct 0x2587810 : Call-ID = 11025
11:42:02.461 SB.CCM : Org Acct = 5558542245 Dst Acct = T02
11:42:02.462 SB.CCM : Org Port ID = 0/2.0 Dst Port ID = 0/0.0
11:42:02.462 SB.CCM : Org TID = Fxs Dst TID =
11:42:02.463 SB.CCM isMappable: Call Connection Type is TDM_TO_RTP
11:42:02.464 SB.CCM isMappable: Reserving RTP Channel 0/1.1
11:42:02.465 SB.CCM isMappable: Creating SDP Offer
11:42:02.468 SB.CCM updateOfferWithEndpointConfig: DTMF(NTE 101), VAD(off), ptime(5)
11:42:02.470 SB.CCM translateOffer: offer codec list: PCMU
11:42:02.470 SB.CCM translateOffer: revised offer codec list: PCMU
11:42:02.472 SB.CCM translateOffer: codec list after answerer: PCMU
11:42:02.475 SB.CCM translateOffer: DTMF signaling: answerer has no restrictions configured, passing offer(NTE 101) through
11:42:02.476 SB.CCM translateOffer: success
11:42:02 MEDIA.ANCHORING MANAGER Allocating media port.
11:42:02 MEDIA.ANCHORING ANCHORING getSubstitutePort: No matching callIdMap entry found for call 11025
11:42:02 MEDIA.ANCHORING ANCHORING Call ID map : Added new entry : call ID 11025 : session -1377186122INIP4127.0.0.3 : version 1 : index 2442
11:42:02 MEDIA.ANCHORING ANCHORING New media entry : type(5), callID(11025), sessionID(-1377186122INIP4127.0.0.3), original IP(127.0.0.3) ports(12442-12443), substitute IP(0.0.0.0) ports(12442-12443), RtpChannel(0/1.1), connection(0x2534d10), sdpOverride(5), me(0x3426310). RtpChannel 0/1.1
11:42:02.481 SB.CALL 11025 Idle Call sent from 5558542245 to T02 (5553962120)
Something really strange is going on. The phone is sending 706-396-2120 but the digits are being processed as 555-396-2120 internally. Do you also have a voice user with the exact pattern of either 706-396-2120 or 555-396-2120 ?
Post the complete configuration sans passwords, please.
Jayh,
No I missed changing that 706 to 555 when I was trying to scrub the configuration. Since I messed that up anyways heres the actual log for the call:
11:41:57.954 PM.0:2 Idle Processed OFFHOOK
11:41:57.955 PM.0:2 State change >> Idle->Requesting Dialtone
11:41:57.956 SA.7068542245 Ca:0 Idle rcvd: AcctPhoneMgr_appearance(ON) from PM
11:41:57.956 SA.7068542245 Ca:0 Idle State change >> Idle->DigitGathering (CAS_ReqDigits)
11:41:57.957 SA.7068542245 Ca:0 DigitGathering sent: AcctPhoneMgr_cachg(CAS_ReqDigits) to PM
11:41:57.957 PM.0:2 Requesting Dialtone CACHG:ReqDigits on primary CA
11:41:57.958 PM.0:2 State change >> Requesting Dialtone->SendingDigits
11:41:57.961 TONESERVICES.EVENTS fxs 0/2 - empty - Tone Detection: Request resource
11:41:57.961 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: DSP channel allocated for the resource
11:41:57.962 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: constructed
11:41:57.963 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: starting
11:41:57.963 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: TDM map
11:41:58.462 TONESERVICES.EVENTS fxs 0/2 - empty - DialTone Generation: Request resource
11:41:58.463 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: DSP channel allocated for the resource
11:41:58.463 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: constructed
11:41:58.464 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: starting
11:41:58.465 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: TDM map
11:42:00.628 PM.0:2 SendingDigits Digit 7 processed
11:42:00.629 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (7) event
11:42:00.630 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: stopping
11:42:00.630 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: TDM unmap
11:42:00.631 RTP.CHANNEL fxs 0/2 - dsp 0/1.1 - DialTone Generation: releasing RTP resource
11:42:00.632 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - DialTone Generation: release
11:42:00.632 SA.7068542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(7) from PM
11:42:00.633 SA.7068542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:00.837 PM.0:2 SendingDigits Digit 0 processed
11:42:00.839 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (0) event
11:42:00.839 SA.7068542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(0) from PM
11:42:00.840 SA.7068542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:01.027 PM.0:2 SendingDigits Digit 6 processed
11:42:01.028 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (6) event
11:42:01.029 SA.7068542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(6) from PM
11:42:01.029 SA.7068542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:01.227 PM.0:2 SendingDigits Digit 3 processed
11:42:01.228 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (3) event
11:42:01.229 SA.7068542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(3) from PM
11:42:01.229 SA.7068542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:01.437 PM.0:2 SendingDigits Digit 9 processed
11:42:01.439 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (9) event
11:42:01.439 SA.7068542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(9) from PM
11:42:01.440 SA.7068542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:01.637 PM.0:2 SendingDigits Digit 6 processed
11:42:01.639 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (6) event
11:42:01.639 SA.7068542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(6) from PM
11:42:01.640 SA.7068542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:01.837 PM.0:2 SendingDigits Digit 2 processed
11:42:01.839 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (2) event
11:42:01.839 SA.7068542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(2) from PM
11:42:01.840 SA.7068542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:02.047 PM.0:2 SendingDigits Digit 1 processed
11:42:02.048 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (1) event
11:42:02.049 SA.7068542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(1) from PM
11:42:02.050 SA.7068542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:02.248 PM.0:2 SendingDigits Digit 2 processed
11:42:02.248 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (2) event
11:42:02.249 SA.7068542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(2) from PM
11:42:02.250 SA.7068542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:02.448 PM.0:2 SendingDigits Digit 0 processed
11:42:02.448 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: received digit (0) event
11:42:02.449 SA.7068542245 Ca:0 DigitGathering rcvd: AcctPhoneMgr_dialDigit(0) from PM
11:42:02.450 SA.7068542245 Ca:0 DigitGathering Named-digit-timeout waiting 0 seconds for more digits
11:42:02.450 SA.7068542245 Ca:0 DigitGathering State change >> DigitGathering->DigitGathering (CAS_Active)
11:42:02.451 SA.7068542245 Ca:0 DigitGathering sent: AcctPhoneMgr_cachg(CAS_Active) to PM
11:42:02.452 PM.0:2 State change >> SendingDigits->Call Pending
11:42:02.454 SA.7068542245 Ca:0 DigitGathering sent: call to SB
11:42:02.454 SA.7068542245 Ca:0 DigitGathering State change >> DigitGathering->CallPending (CAS_Active)
11:42:02.455 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: stopping
11:42:02.456 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: TDM unmap
11:42:02.456 RTP.CHANNEL fxs 0/2 - dsp 0/1.1 - Tone Detection: releasing RTP resource
11:42:02.457 TONESERVICES.EVENTS fxs 0/2 - dsp 0/1.1 - Tone Detection: release
11:42:02.458 SA.7068542245 rcvd: AcctPhoneMgr_billingCode from PM
11:42:02.459 SB.CALL 11025 Idle Called the call routine with 7063962120
11:42:02 SB.TGMgr For dialed number 7063962120, against template NXX-NXX-XXXX, on TrunkGroup SIP, the score is 1000
11:42:02.460 SB.CCM isMappable:
11:42:02.461 SB.CCM : Call Struct 0x2587810 : Call-ID = 11025
11:42:02.461 SB.CCM : Org Acct = 7068542245 Dst Acct = T02
11:42:02.462 SB.CCM : Org Port ID = 0/2.0 Dst Port ID = 0/0.0
11:42:02.462 SB.CCM : Org TID = Fxs Dst TID =
11:42:02.463 SB.CCM isMappable: Call Connection Type is TDM_TO_RTP
11:42:02.464 SB.CCM isMappable: Reserving RTP Channel 0/1.1
11:42:02.465 SB.CCM isMappable: Creating SDP Offer
11:42:02.468 SB.CCM updateOfferWithEndpointConfig: DTMF(NTE 101), VAD(off), ptime(0)
11:42:02.470 SB.CCM translateOffer: offer codec list: PCMU
11:42:02.470 SB.CCM translateOffer: revised offer codec list: PCMU
11:42:02.472 SB.CCM translateOffer: codec list after answerer: PCMU
11:42:02.475 SB.CCM translateOffer: DTMF signaling: answerer has no restrictions configured, passing offer(NTE 101) through
11:42:02.476 SB.CCM translateOffer: success
11:42:02 MEDIA.ANCHORING MANAGER Allocating media port.
11:42:02 MEDIA.ANCHORING ANCHORING getSubstitutePort: No matching callIdMap entry found for call 11025
11:42:02 MEDIA.ANCHORING ANCHORING Call ID map : Added new entry : call ID 11025 : session -1377186122INIP4127.0.0.3 : version 1 : index 2442
11:42:02 MEDIA.ANCHORING ANCHORING New media entry : type(0), callID(11025), sessionID(-1377186122INIP4127.0.0.3), original IP(127.0.0.3) ports(12442-12443), substitute IP(0.0.0.0) ports(12442-12443), RtpChannel(0/1.1), connection(0x2534d10), sdpOverride(0), me(0x3426310). RtpChannel 0/1.1
11:42:02.481 SB.CALL 11025 Idle Call sent from 7068542245 to T02 (7063962120)
OK that makes sense.
Do you have something like this in your configuration:
voice user 7063962120
connect fxs 0/2
first-name "Millie"
last-name "Watt"
!
interface fxs 0/2
no shutdown
!
And if you do a :
show int fxs 0/2
(or whatever fxs to which the 7063962120 user is connected)
Do you see something like:
fxs 0/2 is UP
Two-wire Status is: Onhook
Busyout Status is: Not Configured
Test Status is INACTIVE
No Tests
Here's the config minus som of the management settings and IP addresses, and password fields:
hostname "
enable password encrypted
!
clock timezone -5-Eastern-Time
!
ip subnet-zero
ip classless
ip routing
!
!
!
!
!
event-history on
logging forwarding on
logging facility local7
logging forwarding priority-level smdr
no logging email
!
service password-encryption
!
!
!
ip firewall
no ip firewall alg msn
no ip firewall alg mszone
no ip firewall alg h323
!
!
!
!
!
no dot11ap access-point-control
!
!
!
!
!
!
!
!
!
!
!
qos map ConfigWizardQoSMap 20
match dscp 46
priority 2000
set dscp 46
set cos 7
!
!
!
ip flow top-talkers
top 8
!
interface eth 0/1
encapsulation 802.1q
no shutdown
!
interface eth 0/1.1
vlan-id 1 native
ip address 255.255.0.0
no shutdown
interface eth 0/1.1004
vlan-id 1004
ip address 255.255.0.0
media-gateway ip primary
no shutdown
!
!
!
interface t1 0/1
shutdown
!
interface t1 0/2
description PRI
lbo short 31
tdm-group 1 timeslots 1-24 speed 64
no shutdown
!
!
interface pri 1
isdn switch-type 5ess
isdn name-delivery proceeding
connect t1 0/2 tdm-group 1
digits-transferred 4
no shutdown
!
!
interface fxs 0/1
no shutdown
!
interface fxs 0/2
no shutdown
!
interface fxs 0/3
no shutdown
!
interface fxs 0/4
no shutdown
!
isdn-group 1
min-channels 1
connect pri 1
!
!
!
!
timing-source t1 0/2
!
timing-source internal secondary
!
!
!
!
!
!
no tftp server
no tftp server overwrite
http server
http secure-server
no snmp agent
no ip ftp server
no ip scp server
no ip sntp server
!
ip sip
ip sip udp 5060
no ip sip tcp
!
!
!
voice feature-mode network
voice forward-mode local
!
!
!
!
!
!
!
voice dial-plan 1 local NXX-NXX-XXXX
voice dial-plan 2 long-distance 1-NXX-NXX-XXXX
voice dial-plan 3 toll-free 1-8XX-NXX-XXXX
voice dial-plan 4 international 011-$
voice dial-plan 5 extensions 7063962115
voice dial-plan 6 extensions 7063962120
voice dial-plan 7 extensions 7068542245
voice dial-plan 8 extensions 7068542930
!
!
!
!
voice codec-list fax
default
codec g711ulaw
!
!
!
voice trunk T01 type isdn
description "PRI"
resource-selection circular descending
caller-id-override number-inbound 7068542160 if-no-cpn
caller-id-override emergency-outbound 7068542160
connect isdn-group 1
alc
modem-passthrough
rtp delay-mode adaptive
rtp packet-delay fax 150
rtp packet-delay maximum 120
!
voice trunk T02 type sip
sip-server primary
!
!
voice grouped-trunk SIP
description "FPOE"
trunk T02
accept $ cost 0
accept NXX-NXX-XXXX cost 0
accept 1-NXX-NXX-XXXX cost 0
accept 1-800-NXX-XXXX cost 0
accept 1-888-NXX-XXXX cost 0
accept 1-877-NXX-XXXX cost 0
accept 1-866-NXX-XXXX cost 0
accept 1-855-NXX-XXXX cost 0
accept 1-900-NXX-XXXX cost 0
accept 1-976-NXX-XXXX cost 0
accept NXX-976-XXXX cost 0
accept 011-$ cost 0
accept 411 cost 0
accept 611 cost 0
accept 911 cost 0
accept 0-NXX-NXX-XXXX cost 0
accept 10-10-XXX-$ cost 0
!
!
voice grouped-trunk PRI
trunk T01
accept 7063962116 cost 0
accept 7063962117 cost 0
accept 7063962118 cost 0
accept 7063962119 cost 0
accept 7063962121 cost 0
accept 7063962122 cost 0
accept 7063962123 cost 0
accept 7063962124 cost 0
accept 7063962125 cost 0
accept 7063962126 cost 0
accept 7063962127 cost 0
accept 7063962128 cost 0
accept 7063962129 cost 0
accept 7068542128 cost 0
accept 7068542162 cost 0
accept 7068542170 cost 0
accept 7068542171 cost 0
accept 7068542183 cost 0
accept 7068542247 cost 0
accept 7068542248 cost 0
accept 7068542251 cost 0
accept 7068542253 cost 0
accept 7068542254 cost 0
accept 7068542256 cost 0
accept 7068542262 cost 0
accept 7068542263 cost 0
accept 7068542266 cost 0
accept 7063962075 cost 0
accept 7063962076 cost 0
accept 7063962077 cost 0
accept 7063962078 cost 0
accept 7063962079 cost 0
accept 7063962080 cost 0
accept 7063962081 cost 0
accept 7063962082 cost 0
accept 7063962083 cost 0
accept 7063962084 cost 0
accept 7063962110 cost 0
accept 7063962111 cost 0
accept 7063962112 cost 0
accept 7063962113 cost 0
accept 7063962114 cost 0
accept 7068542160 cost 0
!
!
voice user 7063962115
connect fxs 0/3
no call-waiting
modem-passthrough
no echo-cancellation
rtp packet-delay fax 150
codec-list fax
!
!
voice user 7063962120
connect fxs 0/4
no call-waiting
modem-passthrough
rtp packet-delay fax 150
codec-list fax
!
!
voice user 7068542245
connect fxs 0/2
no call-waiting
modem-passthrough
rtp packet-delay fax 150
codec-list fax
!
voice user 7068542930
connect fxs 0/1
no call-waiting
modem-passthrough
rtp packet-delay fax 150
codec-list fax
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
ip sip grammar from host domain
ip sip grammar to host domain
!
ip sip qos dscp 46
!
!
!
ip rtp symmetric-filter
!
I had edited the orginal post. After I posted I relized it was a seperate simultaneous call that was going on.
Configuration looks good. What does "show interface fxs 0/4" show? Can the device connected there make outbound calls?
I would remove the following, they didn't fix it and may cause problems:
voice dial-plan 5 extensions 7063962115
voice dial-plan 6 extensions 7063962120
voice dial-plan 7 extensions 7068542245
voice dial-plan 8 extensions 7068542930
Also these are potential abuse/fraud vectors, you might want to change to reject:
accept 1-900-NXX-XXXX cost 0
accept 1-976-NXX-XXXX cost 0
accept NXX-976-XXXX cost 0
Another possibility is ring-trip. If the pair connected to fxs 0/4 has a partial short it will answer and immediately drop. Try unplugging the bridge clips at the punch block closest to the Adtran device. Do calls to all of the analog ports act the same way?
Jwable,
Thanks for posting! Our officially supported application with this product is to send all calls out to a SIP server. In this supported scenario, the only time that a call would pass directly between ports locally is if the SIP server is unreachable. However, you could try using specific "reject" statements on all your grouped trunks for the dialed number. This may get you your desired behavior. However, since the call isn't reaching a SIP server, SIP PBX, or Feature Server the call features may be limited once the call is connected.
Thanks!
David
David,
Thanks that clears it up. The problem with trying the reject statements is there legitimate calls to and from the outside world as well. Unless there is an option for a reject statement that allows you specify reject calls only from a particular voice user to another particular voice user on the SIP trunk allowing the call to then complete via FXS to FXS trunk.
John Wable
Well, I stand very corrected! My understanding was that the most specific match of a digit pattern always won.
jwable wrote:
Thanks that clears it up. The problem with trying the reject statements is there legitimate calls to and from the outside world as well. Unless there is an option for a reject statement that allows you specify reject calls only from a particular voice user to another particular voice user on the SIP trunk allowing the call to then complete via FXS to FXS trunk.
A reject statement for the number of a locally-connected voice user on the SIP grouped-trunk shouldn't cause an issue. Anyone originating a call to that pattern from behind the Adtran will connect locally which is what you are trying to accomplish.
The reject statement applies outbound on the grouped-trunk, so someone calling in on the SIP trunk from outside won't be affected and the call will still complete to the local voice user.