cancel
Showing results for 
Show  only  | Search instead for 
Did you mean: 
Anonymous
Not applicable

FXS to FXS Call

Jump to solution

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

Labels (1)
0 Kudos
1 Solution

Accepted Solutions
jayh
Honored Contributor
Honored Contributor

Re: FXS to FXS Call

Jump to solution

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.

View solution in original post

0 Kudos
22 Replies
jayh
Honored Contributor
Honored Contributor

Re: FXS to FXS Call

Jump to solution

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.

Anonymous
Not applicable

Re: FXS to FXS Call

Jump to solution

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

jayh
Honored Contributor
Honored Contributor

Re: FXS to FXS Call

Jump to solution

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: 

  • Voice user digit string (and/or alias)
  • Exactly what is dialed by the caller
  • Matching dial-plan entry under "voice dial-plan" such as: "voice dial-plan 1 local NXX-XXXX" for 7-digit or "voice dial-plan 1 local NXX-NXX-XXXX" for ten-digit.
Anonymous
Not applicable

Re: FXS to FXS Call

Jump to solution

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

jayh
Honored Contributor
Honored Contributor

Re: FXS to FXS Call

Jump to solution

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.

Anonymous
Not applicable

Re: FXS to FXS Call

Jump to solution

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.

jayh
Honored Contributor
Honored Contributor

Re: FXS to FXS Call

Jump to solution

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.  

Anonymous
Not applicable

Re: FXS to FXS Call

Jump to solution

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

jayh
Honored Contributor
Honored Contributor

Re: FXS to FXS Call

Jump to solution

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.

Anonymous
Not applicable

Re: FXS to FXS Call

Jump to solution

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)

jayh
Honored Contributor
Honored Contributor

Re: FXS to FXS Call

Jump to solution

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.

Anonymous
Not applicable

Re: FXS to FXS Call

Jump to solution

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)

jayh
Honored Contributor
Honored Contributor

Re: FXS to FXS Call

Jump to solution

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

Anonymous
Not applicable

Re: FXS to FXS Call

Jump to solution

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

!

Anonymous
Not applicable

Re: FXS to FXS Call

Jump to solution

I had edited the orginal post.  After I posted I relized it was a seperate simultaneous call that was going on.

jayh
Honored Contributor
Honored Contributor

Re: FXS to FXS Call

Jump to solution

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? 

Anonymous
Not applicable

Re: FXS to FXS Call

Jump to solution

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

Anonymous
Not applicable

Re: FXS to FXS Call

Jump to solution

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

jayh
Honored Contributor
Honored Contributor

Re: FXS to FXS Call

Jump to solution

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.