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New Contributor II

SIP to SIP Gateway

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Hello,

I am having an issue with incoming DID routing for a SIP to SIP configuration. I am pretty sure a proxy, either stateful or transparent, is not the answer as I want the Adtran to perform ANI (caller ID) replacement and Emergency CLID override almost exclusively as I don't really need other options. I have a PRI setup in my config but am not using at the moment due to testing the trunks and caller id manipulation, and PRI is currently in production on my PBX. My provider is running me through their PBX for my SIP trunks, effectively making them a transparent RTP proxy, so NAT is not an issue. Port 5060 is being forwarded to the Adtran (10.2.0.145). What I am trying to accomplish is:

Avaya IPO PBX (SIP) <-------> (SIP) Adtran 908e (SIP) <------->  Cisco Meraki Firewall <-------> (SIP) Provider/PSTN

Currently, outbound SIP calling works and ANI replacement is working correctly. What is not working is inbound DID to PBX via SIP. I have done just provider to Adtran to PRI setup which worked with inbound and outbound calls:

Avaya IPO PBX (PRI) <-------> (PRI) Adtran 908e (SIP) <------->  Cisco Meraki Firewall <-------> (SIP) Provider/PSTN

All * are there to hide my phone numbers, outside IPs, and CLID names.

hostname "TA908e"

enable password encrypted 141c53aaf81472e01cf087537a57e90a8b95

!

!

clock timezone -7-Mountain-Time

!

ip subnet-zero

ip classless

ip default-gateway 10.2.0.1

ip routing

ipv6 unicast-routing

!

!

domain-name "*"

name-server 10.2.0.10 10.0.0.10

!

!

no auto-config

auto-config authname adtran encrypted password 333572c999626609298cc1e122bfb0cf17ea

!

event-history on

no logging forwarding

no logging email

!

service password-encryption

!

username "admin" password encrypted "2921684f70c8e5f65910e17ad71b507967cd"

!

banner motd #

        Important

    

The ethernet 0/1 interface is enabled with an address of 10.2.0.145

Telnet/SSH access is also enabled.

#

!

!

no ip firewall alg msn

no ip firewall alg mszone

no ip firewall alg h323

!

!

!

!

!

!

!

!

no dot11ap access-point-control

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

interface eth 0/1

  ip address  10.2.0.145  255.255.255.0

  no shutdown

  media-gateway ip primary

!

!

interface eth 0/2

  no ip address

  shutdown

!

!

!

!

interface t1 0/1

  shutdown

!

interface t1 0/2

  shutdown

!

interface t1 0/3

  tdm-group 1 timeslots 1-24 speed 64

  no shutdown

!

interface t1 0/4

  shutdown

!

!

interface pri 1

  connect t1 0/3 tdm-group 1

  no shutdown

!

!

interface fxs 0/1

  shutdown

!

interface fxs 0/2

  shutdown

!

interface fxs 0/3

  shutdown

!

interface fxs 0/4

  shutdown

!

interface fxs 0/5

  shutdown

!

interface fxs 0/6

  shutdown

!

interface fxs 0/7

  shutdown

!

interface fxs 0/8

  shutdown

!

!

interface fxo 0/0

  shutdown

!

!

isdn-group 1

  connect pri 1

!

!

!

!

!

!

!

!

!

!

!

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

!

!

!

!

!

!

!

!

sip

sip udp 5060

no sip tcp

!

!

!

voice feature-mode network

voice forward-mode network

!

!

!

!

!

!

!

!

voice dial-plan 1 local NXX-NXX-XXXX

voice dial-plan 2 local 1-NXX-NXX-XXXX

!

!

!

!

voice codec-list DEFAULT

  default

  codec g711ulaw

!

!

!

voice trunk T01 type sip

  description "AxisVoIP"

  caller-id-override emergency-outbound 1******5555

  match dnis "911" substitute "1******5555" name "C***** W****"

  match ani "5560" substitute "1******5560" name "C***** W**** Enterta"

  match ani "5561" substitute "1******5561" name "C***** W**** Enterta"

  match ani "1******5560" substitute "1******5560" name "C***** W**** Enterta"

  match ani "1******5561" substitute "1******5561" name "C***** W**** Enterta"

  match ani "55XX" substitute "1******55XX" name "CW World HQ"

  match ani "1******55XX" substitute "1******55XX" name "CW World HQ"

  match dnis "911" replace ani "911" name "Emergency"

  sip-server primary **.**.**.119

  codec-list DEFAULT both

!

voice trunk T02 type isdn

  description "PRI to PBX"

  resource-selection circular descending

  connect isdn-group 1

  rtp delay-mode adaptive

!

voice trunk T03 type sip

  description "SIP to PBX"

  sip-server primary 10.2.5.145

  codec-list DEFAULT both

!

!

voice grouped-trunk PRI

  description "ISDN to PBX PRI"

  trunk T02

  accept $ cost 0

!

!

voice grouped-trunk SIP

  description "Trunk to AxisVoIP"

  resource-selection circular

  trunk T01

  accept $ cost 0

!

!

voice grouped-trunk SIPTOPBX

  description "Trunk to PBX"

  trunk T03

  accept $ cost 0

!

!

!

!

!

!

!

!

!

!

!

!

no sip registrar authenticate

!

!

!

!

!

!

!

!

!

!

!

sip qos dscp 46

!

!

!

!

sdp grammar hold rfc3264

!

!

line con 0

  login

  password encrypted 2b22b2a47eb9da840f71cdfea595a8aae24a

!

line telnet 0 4

  login

  password encrypted 3f36269652fad10bc6ae74e13889ea53fb63

  shutdown

line ssh 0 4

  login local-userlist

  no shutdown

!

sntp server 3.north-america.pool.ntp.org version 3

!

!

!

!

end

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New Contributor II

Re: SIP to SIP Gateway

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You are correct that the PBX was rejecting the call. I was forwarding the call route on my Avaya IP office to a dead extension that I though was another active one. I am marking this as answered as I contacted ADTRAN support which helped me further. It turns out that what I was trying to accomplish is not possible without an eSBC license. You need a session boarder controller to route sip to sip trunks. The ADTRAN tech was amazed that my sip to sip calls were even going out. The SBC is needed to route RTP traffic properly as well. What I ended up doing is just using PRI, which my PBX is already configured for. Incoming DIDs are now routing to PBX on PRI. I am using "accecpt $ cost 0" on both SIP trunk and PRI trunk as my provider will handle which numbers are allowed to reach my PBX. I am also using reject statements on PRI along with the accept $, allowing me to route individual numbers to the FXS ports for analog extensions.

View solution in original post

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New Contributor II

Re: SIP to SIP Gateway

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I am also receiving this message in the CLI. Google has yet to reveal an answer...

2018.06.01 14:59:24 SIP.STACK ERROR  MSGBUILDER   SIP Pre-Parser Error (UDP) :

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New Contributor II

Re: SIP to SIP Gateway

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This May be a different topic but thought to post the results to a debug anyway:

debug command: debug sip stack level errors

TA908e#debug sip stack level errors

TA908e#

15:32:24.464 SIP.STACK ERROR  MSGBUILDER   TransportMsgBuilderUdpPrepare - Message preparing failed

15:32:24.464 SIP.STACK ERROR  MSGBUILDER   TransportMsgBuilderReceivedMsg - failed to build (UDP) message - status = -3

2018.06.01 15:32:24 SIP.STACK ERROR  MSGBUILDER   SIP Pre-Parser Error (UDP) :

15:32:27.213 SIP.STACK ERROR  TRANSACTION  RvSipTransactionRespond - Transaction 0x52392f8: Failed - can't send response in state Server General Final Response Sent

15:32:38.471 SIP.STACK ERROR  TRANSACTION  RvSipTransactionRespond - Transaction 0x52390f0: Failed - can't send response in state Terminated

15:32:44.464 SIP.STACK ERROR  MSGBUILDER   TransportMsgBuilderUdpPrepare - Message preparing failed

15:32:44.464 SIP.STACK ERROR  MSGBUILDER   TransportMsgBuilderReceivedMsg - failed to build (UDP) message - status = -3

2018.06.01 15:32:44 SIP.STACK ERROR  MSGBUILDER   SIP Pre-Parser Error (UDP) :

15:32:59.215 SIP.STACK ERROR  TRANSACTION  RvSipTransactionRespond - Transaction 0x52392f8: Failed - can't send response in state Terminated

15:33:04.465 SIP.STACK ERROR  MSGBUILDER   TransportMsgBuilderUdpPrepare - Message preparing failed

15:33:04.465 SIP.STACK ERROR  MSGBUILDER   TransportMsgBuilderReceivedMsg - failed to build (UDP) message - status = -3

2018.06.01 15:33:04 SIP.STACK ERROR  MSGBUILDER   SIP Pre-Parser Error (UDP) :

15:33:06.512 SIP.STACK ERROR  TRANSACTION  RvSipTransactionRespond - Transaction 0x5239500: Failed - can't send response in state Server General Final Response Sent

15:33:24.464 SIP.STACK ERROR  MSGBUILDER   TransportMsgBuilderUdpPrepare - Message preparing failed

15:33:24.464 SIP.STACK ERROR  MSGBUILDER   TransportMsgBuilderReceivedMsg - failed to build (UDP) message - status = -3

2018.06.01 15:33:24 SIP.STACK ERROR  MSGBUILDER   SIP Pre-Parser Error (UDP) :

15:33:27.207 SIP.STACK ERROR  TRANSACTION  RvSipTransactionRespond - Transaction 0x5239708: Failed - can't send response in state Server General Final Response Sent

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Honored Contributor
Honored Contributor

Re: SIP to SIP Gateway

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Your three grouped-trunks all are set to "accept $ cost 0". You'll need to be specific on at least your grouped-trunk PRI and grouped-trunk SIPTOPBX as to what patterns of numbers route to each PBX. By default the TA900 will send inbound SIP to outbound PRI. The pattern matching should be as the number is received by the TA908 before the outgoing trunk does any digit match/substitute.

"Debug voice switchboard" or the noisier "debug voice verbose" might be useful tools.

The pre-parser errors are due to a device sending malformed or non-standard SIP messages to the TA908. This is most commonly associated with softphones but there may be something in the SIP PBX that is sending wonky or proprietary SIP messages that the Adtran can't parse. It's usually just cosmetic. "no events" on the command-line will suppress the noise while configuring.

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New Contributor II

Re: SIP to SIP Gateway

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Ok, I have change the following. I have also put the ADTRAN on a direct outside IP. Following the changes is the verbose output to where the call is being routed. Attached is the config.

{

voice grouped-trunk PRI

  description "ISDN to PBX PRI"

  trunk T02

  accept 1XXXXXX552[2-9] cost 0

  accept 1XXXXXX55[3,5,6][0-9] cost 0

  accept 1XXXXXX554[0-1] cost 0

  accept 1XXXXXX557[0-4] cost 0

  reject 1XXXXXX6766

!

!

voice grouped-trunk SIPTOPBX

  description "Trunk to PBX"

  trunk T03

  accept 1XXXXXX6766 cost 0

!

!

voice grouped-trunk SIP

  description "Trunk to AxisVoIP"

  trunk T01

  accept $ cost 0

}

{

19:37:50.375 TM.T01 01 SipTM_Idle          rcvd SIP call-leg request: INVITE

19:37:50.376 TM.T01 01 SipTM_Idle          call-leg -> Offering

19:37:50.376 TM.T01 01 SipTM_Idle          State change      >> SipTM_Idle->SipTM_Trying

19:37:50.377 TM.T01 01 SipTM_Trying        SDP offer is not loopback request

19:37:50.377 TM.T01 01 SipTM_Trying        Ignoring P-Asserted-Identity header.

19:37:50.378 TM.T01 01 SipTM_Trying        Processing From for Caller-ID.

19:37:50.378 TM.T01 01 SipTM_Trying        Caller ID Name  = "MY,Name"

19:37:50.378 TM.T01 01 SipTM_Trying        Caller ID Number = "1XXXXXXXXXX"

19:37:50.378 TM.T01 01 SipTM_Trying        info: unable to set redirect number(s) from INVITE

19:37:50.379 TM.T01 01 SipTM_Trying        sent: TA->InboundCall

19:37:50.379 TM.T01 01 Looking up source address for destination XX.XXX.XXX.119

19:37:50.379 TM.T01 01 call-leg (0x52ac0e0) -> src: XX.XX.XX.186 : 5060  dst: XX.XXX.XXX.119 : 5060

19:37:50.381 TM.T01 01 SipTM_Trying        sent: 100 Trying

19:37:50.381 TA.T01 01 TAIdle              rcvd: inboundCall from TM

19:37:50.381 TA.T01 01 State change      >> TAIdle->TAInboundCall (TAS_Calling)

19:37:50.382 TA.T01 01 Failed - DID translation: no match for 1XXXXXX6766, using 1XXXXXX6766

19:37:50.382 TA.T01 01 TAIdle              sent: call to SB

19:37:50.382 TM.T01 01 SipTM_Trying        tachg -> TAInboundCall

19:37:50.383 TM.T01 01 SipTM_Trying        State change      >> SipTM_Trying->SipTM_Pending

19:37:50.383 SB.CALL 1 Idle                Called the call routine with 1XXXXXX6766

19:37:50 SB.TGMgr TrunkGroup PRI rejects number 1XXXXXX6766 based on template 1XXXXXX6766

19:37:50 SB.TGMgr For dialed number 1XXXXXX6766, against template 1XXXXXX6766, on TrunkGroup SIPTOPBX, the score is 12000

19:37:50 SB.TGMgr For dialed number 1XXXXXX6766, against template $, on TrunkGroup SIP, the score is 500

19:37:50.384 SB.CCM isMappable:

19:37:50.384 SB.CCM  :  Call Struct 0x347f010 :  Call-ID = 1

19:37:50.384 SB.CCM  :  Org Acct = T01    Dst Acct = T03

19:37:50.385 SB.CCM  :  Org Port ID = SipTrunk 0/0  Dst Port ID = unknown 0/0

19:37:50.385 SB.CCM  :  SDP Transaction = CallID: 1

19:37:50.385 SB.CCM  :  SDP Offer = 0x03653310, (XX.XXX.XXX.119:15494)

19:37:50.385 SB.CCM isMappable: Call Connection Type is RTP_TO_RTP

19:37:50.385 SB.CCM handleRtpToRtp: Modifying SDP Offer

19:37:50.386 SB.CCM translateOffer: offer codec list: PCMU PCMA GSM

19:37:50.387 SB.CCM translateOffer: revised offer codec list: PCMU

19:37:50.387 SB.CCM translateOffer: codec list after answerer: PCMU

19:37:50.388 SB.CCM translateOffer: DTMF signaling: answerer has no restrictions configured, passing offer(NTE 101) through

19:37:50.388 SB.CCM translateOffer: success

19:37:50.388 MEDIA.MANAGER Allocating media port.

19:37:50.388 MEDIA.MANAGER getSubstitutePort: No matching callIdMap entry found for call 1

19:37:50.389 MEDIA.MANAGER Call ID map : Added new entry : call ID 1 : session root266568222INIP4XX.XXX.XXX.119 : version 266568222 : index 0

19:37:50.389 MEDIA.MANAGER New media entry : type(0), callID(1), sessionID(root266568222INIP4XX.XXX.XXX.119), original IP(XX.XXX.XXX.119) ports(15494-15495), substitute IP(::) ports(10000-10001), RtpChannel(NULL), connection(0x3654410), sdpOverride(0), me(0x3654310). No RtpChannel

19:37:50.389 SB.CALL 1 Idle                Call sent from T01 to T03 (1XXXXXX6766)

19:37:50.390 SB.CALL 1 State change      >> Idle->Delivering

19:37:50.390 TA.T01 01 TAInboundCall        CallResp event accepted

19:37:50.390 TA.T01 01 State change      >> TAInboundCall->TAConnectWaitIn (TAS_Calling)

19:37:50.390 TA.T03 100 State change      >> TAIdle->TAOutGoing (TAS_Delivering)

19:37:50.391 TM.T03 100 SipTM_Idle          State change      >> SipTM_Idle->Delivering

19:37:50.391 TM.T03 100 Delivering          Applying E.164 settings to called party number (1XXXXXX6766)

19:37:50.391 TM.T03 100 Delivering          Skipping E.164 conversion due to voice international-prefix setting

19:37:50.391 TM.T03 100 Delivering          Applying E.164 settings to calling party number (17209394155)

19:37:50.392 TM.T03 100 Delivering          From user grammar setting is: domestic

19:37:50.392 TM.T03 100 Delivering          Skipping E.164 conversion due to From user grammar setting

19:37:50.392 TM.T03 100 Looking up source address for destination 10.2.5.145

19:37:50.392 TM.T03 100 call-leg (0x52ac310) -> src: XX.XX.XX.186 : 5060  dst: 10.2.5.145 : 5060

19:37:50.393 TM.T03 100 SDP DPI call ID 1 : No media bin.

19:37:50.394 TM.T03 100 Processing new SDP entries.

19:37:50.394 TM.T03 100 Checking for internal Media Gateway IP Address

19:37:50.394 TM.T03 100 RTP Channel is NULL, Media Gateway must not be involved in call

19:37:50.394 TM.T03 100 Undo of previous operation not required (RTP NAT Entry for XX.XXX.XXX.119:15494 not found)

19:37:50.394 TM.T03 100 Checking for internal Media Gateway IP Address

19:37:50.395 TM.T03 100 Given RTP Channel is null, checking for hairpinned RTP Channel

19:37:50.395 TM.T03 100 RTP Channel is NULL, Media Gateway must not be involved in call

19:37:50.395 TM.T03 100 No action taken, IPv4 firewall is not enabled

19:37:50.397 TM.T03 100 Delivering          call-leg -> Inviting

19:37:50.398 TM.T03 100 Delivering          sent: INVITE

19:37:50.398 SB.CALL 1 Delivering          Called the deliverResponse routine from Delivering

19:37:50.398 SB.CALL 1 Delivering          DeliverResponse(accept) sent from T03 to T01

19:37:50.399 TA.T01 01 TAConnectWaitIn      deliverResponse event accepted

19:37:50.399 TA.T01 01 TAConnectWaitIn      ERROR! deliverResponse ignored

19:37:50 SB.CallStructObserver 1 Created

19:37:50 SB.CallStructObserver 1 <-> 1e65f29e6d1da8822ec7817e5551048c@XX.XXX.XXX.119:5060

19:37:53.399 TM.T03 100 INVITE rollover timeout

19:37:53.399 TM.T03 100 Delivering          Sip_CreateCallLegNextServer with default validator

19:37:53.399 TM.T03 100 Delivering          State change      >> Delivering->SipTM_Closing

19:37:53.402 TM.T03 100 SipTM_Closing        sent: TA->Clear

19:37:53.403 TM.T03 100 SipTM_Closing        call-leg -> Terminated

19:37:53.403 TA.T03 100 TAOutGoing          rcvd: clear from TM

19:37:53.403 TA.T03 100 State change      >> TAOutGoing->TATrunkClearing (TAS_Clearing)

19:37:53.403 TM.T03 100 SipTM_Closing        tachg -> TATrunkClearing

19:37:53.404 TM.T03 100 SipTM_Closing        State change      >> SipTM_Closing->SipTM_Terminated

19:37:53.404 TM.T03 100 SipTM_Terminated    sent: TA->AppearanceOff

19:37:53.404 TM.T03 100 SipTM_Terminated    State change      >> SipTM_Terminated->SipTM_Idle

19:37:53.404 SB.CALL 1 Delivering          Called the clearCall routine

19:37:53.405 SB.CALL 1 Delivering          SIP Proxy rejected call to 1XXXXXX6766 for survivability - no matching Proxy user

19:37:53.405 SB.CALL 1 Delivering          No available resources on call from T01 to T03 (last attempt)

19:37:53.405 SB.CALL 1 State change      >> Delivering->Clearing

19:37:53.405 TA.T03 100 TATrunkClearing      rcvd: appearance off from TM

19:37:53.405 TA.T03 100 State change      >> TATrunkClearing->TAClearingComplete (TAS_Clearing)

19:37:53.405 TA.T03 100 TATrunkClearing      Processing an appearance OFF

19:37:53.406 TA.T01 01 TAConnectWaitIn      ClearCall event accepted

19:37:53.406 TA.T01 01 State change      >> TAConnectWaitIn->TAClearingComplete (TAS_Clearing)

19:37:53.406 TM.T01 01 SipTM_Pending        tachg -> TAClearingComplete

19:37:53.406 TM.T01 01 SipTM_Pending        State change      >> SipTM_Pending->SipTM_CallFail

19:37:53.408 TM.T01 01 SipTM_CallFail      call-leg -> Disconnected

19:37:53.408 TM.T01 01 SipTM_CallFail      CallLegStateChanged to Disconnected - TM change to closing state.

19:37:53.408 TM.T01 01 SipTM_CallFail      State change      >> SipTM_CallFail->SipTM_Closing

19:37:53.408 TM.T01 01 SipTM_Closing        sent: TA->Clear

19:37:53.408 TM.T01 01 SipTM_CallFail      sent: 503

19:37:53.409 TM.T01 01 SipTM_Closing        State change      >> SipTM_Closing->SipTM_Terminated

19:37:53.409 TM.T01 01 SipTM_Terminated    sent: TA->AppearanceOff

19:37:53.409 TM.T01 01 SipTM_Terminated    State change      >> SipTM_Terminated->SipTM_Idle

19:37:53.409 SB.CALL 1 Clearing            Called the clearResponse routine

19:37:53.410 SB.CALL 1 State change      >> Clearing->CallIdlePending

19:37:53.410 SB.CCM release:

19:37:53.410 SB.CCM  :  Call Struct 0x347f010 :  Call-ID = 1

19:37:53.410 SB.CCM  :  Org Acct = T01    Dst Acct = T03

19:37:53.410 SB.CCM  :  Org Port ID = SipTrunk 0/0  Dst Port ID = SipTrunk 0/0.299

19:37:53.410 SB.CCM  :  SDP Transaction = CallID: 1

19:37:53.411 SB.CCM  :  SDP Offer = 0x03653310, (XX.XXX.XXX.119:15494)

19:37:53.411 SB.CCM release: Call Connection Type is RTP_TO_RTP

19:37:53.411 SB.CALL 1 CallIdlePending      ClearResponse sent from T01 to T03

19:37:53.411 TA.T01 01 TAClearingComplete  rcvd: clear from TM

19:37:53.412 TA.T01 01 TAClearingComplete  rcvd: appearance off from TM

19:37:53.412 TA.T01 01 TAClearingComplete  Clear Local Variables

19:37:53.412 TA.T01 01 State change      >> TAClearingComplete->TAIdle (TAS_Idle)

19:37:53.412 TM.T01 01 SipTM_Idle          tachg -> TAIdle

19:37:53.413 TA.T03 100 TAClearingComplete  clearResponse event accepted

19:37:53.413 TA.T03 100 TAClearingComplete  Clear Local Variables

19:37:53.413 TA.T03 100 State change      >> TAClearingComplete->TAIdle (TAS_Idle)

19:37:53.413 TM.T03 100 SipTM_Idle          tachg -> TAIdle

19:37:53 SB.CallStructObserver 1 Finalized

2018.06.05 19:37:54 SMDR 1          06/05/2018 19:37:50      0.0 0    E  00/00 NAME,MY    1XXXXXXXXXX    00/00 T03            1XXXXXX6766    0 N

19:37:56.203 MEDIA.MANAGER Remove Call ID map entry for call 1

}

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Honored Contributor

Re: SIP to SIP Gateway

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Your configuration on the internal interface and PBX needs some tweaks.

!

interface eth 0/1

  ip address  10.2.0.145  255.255.255.0

  no shutdown

!

Add "media-gateway ip primary" to this interface.

!

voice trunk T03 type sip

  description "SIP to PBX"

  sip-server primary 10.2.5.145

  no registrar require-expires

  codec-list Trunk both

!

I don't see any internal routing in your configuration, but 10.2.5.145 is in a different subnet from the 10.2.0.145 interface eth 0/1. Is there an internal router between eth 0/1 and the PBX?

Add the "media-gateway ip primary" line to your eth 0/1 interface, confirm that there's an internal route to 10.2.5.145, and try the call again with "debug sip stack messages" enabled.

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New Contributor II

Re: SIP to SIP Gateway

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My Voice vLAN is 10.2.5.0/24 and is routed to my data vLAN 10.2.0.0/24. I just set up the adtran on the 10.2.0.0/24 subnet for easier config/troubleshooting from my desk and to move back down to the rack where the PBX is to test PRI, but they are routed through my network firewall. I will be adding it to the 10.2.5.0/24 subnet when config is complete. I have added the media-gateway ip primary to the 10.2.5.145 interface but still no joy. I am running an Avaya IP Office 500 v2 if that makes a difference. I really hope I do not have to get eSBCs.

I also just ran a debug voice verbose using PRI and the DID 6766 added to permit, same issue. It can not route the call. Here is that dump

15:29:55.371 TM.T01 01 SipTM_Idle           rcvd SIP call-leg request: INVITE

15:29:55.371 TM.T01 01 SipTM_Idle           call-leg -> Offering

15:29:55.371 TM.T01 01 SipTM_Idle           State change      >> SipTM_Idle->SipTM_Trying

15:29:55.372 TM.T01 01 SipTM_Trying         SDP offer is not loopback request

15:29:55.373 TM.T01 01 SipTM_Trying         Ignoring P-Asserted-Identity header.

15:29:55.373 TM.T01 01 SipTM_Trying         Processing From for Caller-ID.

15:29:55.373 TM.T01 01 SipTM_Trying         Caller ID Name   = "My,Name"

15:29:55.373 TM.T01 01 SipTM_Trying         Caller ID Number = "11234567890"

15:29:55.374 TM.T01 01 SipTM_Trying         info: unable to set redirect number(s) from INVITE

15:29:55.374 TM.T01 01 SipTM_Trying         sent: TA->InboundCall

15:29:55.374 TM.T01 01 Looking up source address for destination XX.XXX.XXX.119

15:29:55.374 TM.T01 01 call-leg (0x52b9ba0) -> src: XX.XX.XX.186 : 5060  dst: XX.XXX.XXX.119 : 5060

15:29:55.376 TM.T01 01 SipTM_Trying         sent: 100 Trying

15:29:55.376 TA.T01 01 TAIdle               rcvd: inboundCall from TM

15:29:55.376 TA.T01 01 State change      >> TAIdle->TAInboundCall (TAS_Calling)

15:29:55.377 TA.T01 01 Failed - DID translation: no match for 1XXXXXX6766, using 1XXXXXX6766

15:29:55.377 TA.T01 01 TAIdle               sent: call to SB

15:29:55.377 TM.T01 01 SipTM_Trying         tachg -> TAInboundCall

15:29:55.378 TM.T01 01 SipTM_Trying         State change      >> SipTM_Trying->SipTM_Pending

15:29:55.378 SB.CALL 3 Idle                 Called the call routine with 1XXXXXX6766

15:29:55 SB.TGMgr For dialed number 1XXXXXX6766, against template 1XXXXXX6766, on TrunkGroup PRI, the score is 12000

15:29:55 SB.TGMgr TrunkGroup SIPTOPBX rejects number 1XXXXXX6766 based on template $

15:29:55 SB.TGMgr For dialed number 1XXXXXX6766, against template 1-NXX-NXX-XXXX, on TrunkGroup SIP, the score is 2000

15:29:55.379 SB.CALL 3 Idle                 No LOCAL station matched dialed number (1XXXXXX6766)

15:29:55.379 SB.CCM isMappable:

15:29:55.379 SB.CCM  :  Call Struct 0x34ae810 :   Call-ID = 3

15:29:55.379 SB.CCM  :  Org Acct = T01    Dst Acct = T02

15:29:55.379 SB.CCM  :  Org Port ID = SipTrunk 0/0   Dst Port ID = unknown 0/0

15:29:55.380 SB.CCM  :  SDP Transaction = CallID: 3

15:29:55.380 SB.CCM  :  SDP Offer = 0x0344bd10, (XX.XXX.XXX.119:11556)

15:29:55.380 SB.CCM isMappable: Call Connection Type is RTP_TO_TDM

15:29:55.381 SB.CCM isMappable: Reserving RTP Channel 0/1.1

15:29:55.382 SB.CCM translateOffer: offer codec list: PCMU PCMA GSM

15:29:55.382 SB.CCM translateOffer: revised offer codec list: PCMU

15:29:55.382 SB.CCM translateOffer: codec list after answerer: PCMU

15:29:55.383 SB.CCM translateOffer: DTMF signaling: answerer has no restrictions configured, passing offer(NTE 101) through

15:29:55.383 SB.CCM translateOffer: success

15:29:55.383 MEDIA.MANAGER Allocating media port.

15:29:55.384 MEDIA.MANAGER getSubstitutePort: No matching callIdMap entry found for call 3

15:29:55.384 MEDIA.MANAGER Call ID map : Added new entry : call ID 3 : session root1468031030INIP4XX.XXX.XXX.119 : version 1468031030 : index 4

15:29:55.384 MEDIA.MANAGER New media entry : type(0), callID(3), sessionID(root1468031030INIP4XX.XXX.XXX.119), original IP(XX.XXX.XXX.119) ports(11556-11557), substitute IP(::) ports(10004-10005), RtpChannel(0/1.1), connection(0x3449210), sdpOverride(0), me(0x3481f10). RtpChannel 0/1.1

15:29:55.385 SB.CALL 3 Idle                 Call sent from T01 to T02 (1XXXXXX6766)

15:29:55.385 SB.CALL 3 State change      >> Idle->Delivering

15:29:55.385 RTP.MANAGER Isdn(Group) 0/ - empty - RTP: Reserve resource

15:29:55.385 RTP.MANAGER Isdn(Group) 0/ - Dsp 0/1.1 - RTP: (null)

15:29:55.386 RTP.PROVIDER unknown - Dsp 0/1.1 - RTP: reserving already allocated RTP channel

15:29:55.386 TA.T01 01 TAInboundCall        CallResp event accepted

15:29:55.386 TA.T01 01 State change      >> TAInboundCall->TAConnectWaitIn (TAS_Calling)

15:29:55.386 TA.T02 01 State change      >> TAIdle->TAOutGoing (TAS_Delivering)

15:29:55.386 TM.T02 01 tachg_Delivering

15:29:55.387 TM.T02 01 IsdnTmStateIdle->IsdnTmStateOutboundDeliver

15:29:55.387 TM.T02 01 IsdnTmStateOutboundDeliver::enter()

15:29:55.388 SB.CALL 3 Delivering           Called the deliverResponse routine from Delivering

15:29:55.388 SB.CALL 3 Delivering           DeliverResponse(accept) sent from T02 to T01

15:29:55.388 TA.T01 01 TAConnectWaitIn      deliverResponse event accepted

15:29:55.388 TA.T01 01 TAConnectWaitIn      ERROR! deliverResponse ignored

15:29:55 SB.CallStructObserver 3 Created

15:29:55 SB.CallStructObserver 3 <-> 317fe14d0e530571260062d5773148fc@XX.XXX.XXX.119:5060

15:29:55.407 TM.T02 01 IsdnTmStateOutboundDeliver - rcvd unexpected CallRelease

15:29:55.408 TM.T02 01 IsdnTmStateOutboundDeliver->IsdnTmStateIdling

15:29:55.408 TM.T02 01 IsdnTmStateIdling::enter()

15:29:55.408 TM.T02 01 IsdnTmStateIdling - clear trunk appearance

15:29:55.408 TM.T02 01 IsdnTmStateIdling - send appearance off

15:29:55.408 TM.T02 01 IsdnTmStateIdling->IsdnTmStateIdle

15:29:55.408 TM.T02 01 IsdnAppearanceChannel::releaseChannel

15:29:55.409 TM.T02 01 IsdnTmStateIdle::enter()

15:29:55.409 TA.T02 01 TAOutGoing           rcvd: clear from TM

15:29:55.409 TA.T02 01 State change      >> TAOutGoing->TATrunkClearing (TAS_Clearing)

15:29:55.409 TM.T02 01 IsdnTmStateIdle::tachgClearing - send appearance off

15:29:55.410 TA.T02 01 TATrunkClearing      rcvd: appearance off from TM

15:29:55.410 TA.T02 01 State change      >> TATrunkClearing->TAClearingComplete (TAS_Clearing)

15:29:55.410 TA.T02 01 TATrunkClearing      Processing an appearance OFF

15:29:55.410 SB.CALL 3 Delivering           Called the clearCall routine

15:29:55.410 SB.CALL 3 Delivering           Clearing due to Trunk Clear Reason on call from T02 to T01

15:29:55.411 SB.CALL 3 State change      >> Delivering->Clearing

15:29:55.411 TA.T02 01 TAClearingComplete   rcvd: appearance off from TM

15:29:55.411 TA.T02 01 TAClearingComplete   Clear Local Variables

15:29:55.411 TA.T02 01 State change      >> TAClearingComplete->TAIdle (TAS_Idle)

15:29:55.411 TA.T01 01 TAConnectWaitIn      ClearCall event accepted

15:29:55.412 TA.T01 01 State change      >> TAConnectWaitIn->TAClearingComplete (TAS_Clearing)

15:29:55.412 TM.T01 01 SipTM_Pending        tachg -> TAClearingComplete

15:29:55.412 TM.T01 01 SipTM_Pending        State change      >> SipTM_Pending->SipTM_CallFail

15:29:55.414 TM.T01 01 SipTM_CallFail       call-leg -> Disconnected

15:29:55.415 TM.T01 01 SipTM_CallFail       CallLegStateChanged to Disconnected - TM change to closing state.

15:29:55.415 TM.T01 01 SipTM_CallFail       State change      >> SipTM_CallFail->SipTM_Closing

15:29:55.415 TM.T01 01 SipTM_Closing        sent: TA->Clear

15:29:55.415 TM.T01 01 SipTM_CallFail       sent: 0

15:29:55.415 TM.T01 01 SipTM_Closing        State change      >> SipTM_Closing->SipTM_Terminated

15:29:55.416 TM.T01 01 SipTM_Terminated     sent: TA->AppearanceOff

15:29:55.416 TM.T01 01 SipTM_Terminated     State change      >> SipTM_Terminated->SipTM_Idle

15:29:55.416 SB.CALL 3 Clearing             Called the clearResponse routine

15:29:55.417 SB.CALL 3 State change      >> Clearing->CallIdlePending

15:29:55.417 SB.CCM release:

15:29:55.417 SB.CCM  :  Call Struct 0x34ae810 :   Call-ID = 3

15:29:55.417 SB.CCM  :  Org Acct = T01    Dst Acct = T02

15:29:55.417 SB.CCM  :  Org Port ID = SipTrunk 0/0   Dst Port ID = Isdn(Group) 0/0

15:29:55.417 SB.CCM  :  SDP Transaction = CallID: 3

15:29:55.418 SB.CCM  :  SDP Offer = 0x0344bd10, (XX.XXX.XXX.119:11556)

15:29:55.418 SB.CCM  :  RTP Channel = 0/1.1

15:29:55.418 SB.CCM release: Call Connection Type is RTP_TO_TDM

15:29:55.418 SB.CCM release: Releasing RTP Channel 0/1.1

15:29:55.419 RTP.CHANNEL RtpChannel::deallocate, status = 2, allocatedForInterface = 0

15:29:55.419 RTP.CHANNEL Channel 0/1.1 released successfully.

15:29:55.419 SB.CALL 3 CallIdlePending      ClearResponse sent from T01 to T02

15:29:55.419 TA.T01 01 TAClearingComplete   rcvd: clear from TM

15:29:55.419 TA.T01 01 TAClearingComplete   rcvd: appearance off from TM

15:29:55.420 TA.T01 01 TAClearingComplete   Clear Local Variables

15:29:55.420 TA.T01 01 State change      >> TAClearingComplete->TAIdle (TAS_Idle)

15:29:55.420 TM.T01 01 SipTM_Idle           tachg -> TAIdle

15:29:55.420 TA.T02 clearResponse event rejected, no matching CallID3

15:29:55.420 RTP.CHANNEL unknown - Dsp 0/1.1 - RTP: releasing RTP resource

15:29:55.421 RTP.CHANNEL unknown - Dsp 0/1.1 - RTP: releasing

15:29:55 SB.CallStructObserver 3 Finalized

2018.06.06 15:29:56 SMDR 3          06/06/2018 15:29:55      0.0 0    E  00/00 My,Name    11234567890     00/00 T02             1XXXXXX6766     0 N

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New Contributor II

Re: SIP to SIP Gateway

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Here is the SIP STACK debug as requested

15:57:14.358 SIP.STACK MSG     Rx: UDP src=XX.XXX.XXX.119:5060 dst=XX.XX.XX.186:5060

15:57:14.358 SIP.STACK MSG         INVITE sip:1XXXXXX6766@XX.XX.XX.186;user=phone SIP/2.0

15:57:14.358 SIP.STACK MSG         Via: SIP/2.0/UDP XX.XXX.XXX.119:5060;branch=z9hG4bK3d7960b9;rport

15:57:14.358 SIP.STACK MSG         Max-Forwards: 70

15:57:14.358 SIP.STACK MSG         From: "NAME,MY" <sip:1XXXXXX4155@XX.XXX.XXX.119>;tag=as2092f99b

15:57:14.358 SIP.STACK MSG         To: <sip:1XXXXXX6766@XX.XX.XX.186;user=phone>

15:57:14.359 SIP.STACK MSG         Contact: <sip:1XXXXXX4155@XX.XXX.XXX.119:5060>

15:57:14.359 SIP.STACK MSG         Call-ID: 60e4bbe0011748054677fbf339e9a872@XX.XXX.XXX.119:5060

15:57:14.359 SIP.STACK MSG         CSeq: 101 INVITE

15:57:14.359 SIP.STACK MSG         User-Agent: Asterisk PBX (ScopServ)

15:57:14.359 SIP.STACK MSG         Date: Wed, 06 Jun 2018 15:57:15 GMT

15:57:14.360 SIP.STACK MSG         Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

15:57:14.360 SIP.STACK MSG         Supported: replaces,timer

15:57:14.360 SIP.STACK MSG         P-Asserted-Identity: "NAME,MY" <sip:1XXXXXX4155@XX.XXX.XXX.119>

15:57:14.360 SIP.STACK MSG         Content-Type: application/sdp

15:57:14.360 SIP.STACK MSG         Content-Length: 289

15:57:14.360 SIP.STACK MSG

15:57:14.361 SIP.STACK MSG         v=0

15:57:14.361 SIP.STACK MSG         o=root 112625346 112625346 IN IP4 XX.XXX.XXX.119

15:57:14.361 SIP.STACK MSG         s=Asterisk PBX 13.17.0

15:57:14.361 SIP.STACK MSG         c=IN IP4 XX.XXX.XXX.119

15:57:14.361 SIP.STACK MSG         t=0 0

15:57:14.362 SIP.STACK MSG         m=audio 12496 RTP/AVP 0 8 3 101

15:57:14.362 SIP.STACK MSG         a=rtpmap:0 PCMU/8000

15:57:14.362 SIP.STACK MSG         a=rtpmap:8 PCMA/8000

15:57:14.362 SIP.STACK MSG         a=rtpmap:3 GSM/8000

15:57:14.362 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

15:57:14.363 SIP.STACK MSG         a=fmtp:101 0-16

15:57:14.363 SIP.STACK MSG         a=maxptime:150

15:57:14.363 SIP.STACK MSG         a=sendrecv

15:57:14.363 SIP.STACK MSG

15:57:14.370 SIP.STACK MSG     Tx: UDP src=XX.XX.XX.186:5060 dst=XX.XXX.XXX.119:5060

15:57:14.370 SIP.STACK MSG         SIP/2.0 100 Trying

15:57:14.370 SIP.STACK MSG         From: "NAME,MY"<sip:1XXXXXX4155@XX.XXX.XXX.119>;tag=as2092f99b

15:57:14.370 SIP.STACK MSG         To: <sip:1XXXXXX6766@XX.XX.XX.186;user=phone>

15:57:14.371 SIP.STACK MSG         Call-ID: 60e4bbe0011748054677fbf339e9a872@XX.XXX.XXX.119:5060

15:57:14.371 SIP.STACK MSG         CSeq: 101 INVITE

15:57:14.371 SIP.STACK MSG         Via: SIP/2.0/UDP XX.XXX.XXX.119:5060;rport=5060;branch=z9hG4bK3d7960b9

15:57:14.371 SIP.STACK MSG         Contact: <sip:1XXXXXX6766@XX.XX.XX.186:5060;transport=UDP>

15:57:14.371 SIP.STACK MSG         Supported: 100rel,replaces

15:57:14.372 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

15:57:14.372 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R13.2.0.E

15:57:14.372 SIP.STACK MSG         Content-Length: 0

15:57:14.372 SIP.STACK MSG

15:57:14.378 SIP.STACK MSG     Tx: UDP src=XX.XX.XX.186:5060 dst=10.2.5.145:5060

15:57:14.378 SIP.STACK MSG         INVITE sip:1XXXXXX6766@10.2.5.145:5060 SIP/2.0

15:57:14.379 SIP.STACK MSG         From: "NAME,MY" <sip:1XXXXXX4155@10.2.5.145:5060;transport=UDP>;tag=525b990-7f000001-13c4-10ed3-4efcc0d8-10ed3

15:57:14.379 SIP.STACK MSG         To: <sip:1XXXXXX6766@10.2.5.145:5060>

15:57:14.379 SIP.STACK MSG         Call-ID: 52bc760-7f000001-13c4-10ed3-4183a890-10ed3@10.2.5.145

15:57:14.379 SIP.STACK MSG         CSeq: 1 INVITE

15:57:14.379 SIP.STACK MSG         Via: SIP/2.0/UDP XX.XX.XX.186:5060;branch=z9hG4bK-10ed3-421ebd2-79e4efaf

15:57:14.380 SIP.STACK MSG         Max-Forwards: 70

15:57:14.380 SIP.STACK MSG         Supported: 100rel,replaces

15:57:14.380 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

15:57:14.380 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R13.2.0.E

15:57:14.380 SIP.STACK MSG         Contact: <sip:1XXXXXX4155@XX.XX.XX.186:5060;transport=UDP>

15:57:14.381 SIP.STACK MSG         Content-Type: application/sdp

15:57:14.381 SIP.STACK MSG         Content-Length: 242

15:57:14.381 SIP.STACK MSG

15:57:14.381 SIP.STACK MSG         v=0

15:57:14.381 SIP.STACK MSG         o=root 112625346 112625346 IN IP4 XX.XXX.XXX.119

15:57:14.382 SIP.STACK MSG         s=Asterisk PBX 13.17.0

15:57:14.382 SIP.STACK MSG         c=IN IP4 XX.XXX.XXX.119

15:57:14.382 SIP.STACK MSG         t=0 0

15:57:14.382 SIP.STACK MSG         m=audio 12496 RTP/AVP 0 101

15:57:14.382 SIP.STACK MSG         a=maxptime:150

15:57:14.383 SIP.STACK MSG         a=sendrecv

15:57:14.383 SIP.STACK MSG         a=rtpmap:0 PCMU/8000

15:57:14.383 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

15:57:14.383 SIP.STACK MSG         a=fmtp:101 0-15

15:57:14.383 SIP.STACK MSG

15:57:14.884 SIP.STACK MSG SIP stack timer retransmit

15:57:14.884 SIP.STACK MSG     Tx: UDP src=XX.XX.XX.186:5060 dst=10.2.5.145:5060

15:57:14.884 SIP.STACK MSG         INVITE sip:1XXXXXX6766@10.2.5.145:5060 SIP/2.0

15:57:14.884 SIP.STACK MSG         From: "NAME,MY" <sip:1XXXXXX4155@10.2.5.145:5060;transport=UDP>;tag=525b990-7f000001-13c4-10ed3-4efcc0d8-10ed3

15:57:14.885 SIP.STACK MSG         To: <sip:1XXXXXX6766@10.2.5.145:5060>

15:57:14.885 SIP.STACK MSG         Call-ID: 52bc760-7f000001-13c4-10ed3-4183a890-10ed3@10.2.5.145

15:57:14.885 SIP.STACK MSG         CSeq: 1 INVITE

15:57:14.885 SIP.STACK MSG         Via: SIP/2.0/UDP XX.XX.XX.186:5060;branch=z9hG4bK-10ed3-421ebd2-79e4efaf

15:57:14.885 SIP.STACK MSG         Max-Forwards: 70

15:57:14.885 SIP.STACK MSG         Supported: 100rel,replaces

15:57:14.886 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

15:57:14.886 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R13.2.0.E

15:57:14.886 SIP.STACK MSG         Contact: <sip:1XXXXXX4155@XX.XX.XX.186:5060;transport=UDP>

15:57:14.886 SIP.STACK MSG         Content-Type: application/sdp

15:57:14.886 SIP.STACK MSG         Content-Length: 242

15:57:14.887 SIP.STACK MSG

15:57:14.887 SIP.STACK MSG         v=0

15:57:14.887 SIP.STACK MSG         o=root 112625346 112625346 IN IP4 XX.XXX.XXX.119

15:57:14.887 SIP.STACK MSG         s=Asterisk PBX 13.17.0

15:57:14.887 SIP.STACK MSG         c=IN IP4 XX.XXX.XXX.119

15:57:14.888 SIP.STACK MSG         t=0 0

15:57:14.888 SIP.STACK MSG         m=audio 12496 RTP/AVP 0 101

15:57:14.888 SIP.STACK MSG         a=maxptime:150

15:57:14.888 SIP.STACK MSG         a=sendrecv

15:57:14.888 SIP.STACK MSG         a=rtpmap:0 PCMU/8000

15:57:14.888 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

15:57:14.889 SIP.STACK MSG         a=fmtp:101 0-15

15:57:14.889 SIP.STACK MSG

15:57:15.889 SIP.STACK MSG SIP stack timer retransmit

15:57:15.889 SIP.STACK MSG     Tx: UDP src=XX.XX.XX.186:5060 dst=10.2.5.145:5060

15:57:15.889 SIP.STACK MSG         INVITE sip:1XXXXXX6766@10.2.5.145:5060 SIP/2.0

15:57:15.889 SIP.STACK MSG         From: "NAME,MY" <sip:1XXXXXX4155@10.2.5.145:5060;transport=UDP>;tag=525b990-7f000001-13c4-10ed3-4efcc0d8-10ed3

15:57:15.889 SIP.STACK MSG         To: <sip:1XXXXXX6766@10.2.5.145:5060>

15:57:15.890 SIP.STACK MSG         Call-ID: 52bc760-7f000001-13c4-10ed3-4183a890-10ed3@10.2.5.145

15:57:15.890 SIP.STACK MSG         CSeq: 1 INVITE

15:57:15.890 SIP.STACK MSG         Via: SIP/2.0/UDP XX.XX.XX.186:5060;branch=z9hG4bK-10ed3-421ebd2-79e4efaf

15:57:15.890 SIP.STACK MSG         Max-Forwards: 70

15:57:15.890 SIP.STACK MSG         Supported: 100rel,replaces

15:57:15.891 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

15:57:15.891 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R13.2.0.E

15:57:15.891 SIP.STACK MSG         Contact: <sip:1XXXXXX4155@XX.XX.XX.186:5060;transport=UDP>

15:57:15.891 SIP.STACK MSG         Content-Type: application/sdp

15:57:15.891 SIP.STACK MSG         Content-Length: 242

15:57:15.891 SIP.STACK MSG

15:57:15.892 SIP.STACK MSG         v=0

15:57:15.892 SIP.STACK MSG         o=root 112625346 112625346 IN IP4 XX.XXX.XXX.119

15:57:15.892 SIP.STACK MSG         s=Asterisk PBX 13.17.0

15:57:15.892 SIP.STACK MSG         c=IN IP4 XX.XXX.XXX.119

15:57:15.892 SIP.STACK MSG         t=0 0

15:57:15.893 SIP.STACK MSG         m=audio 12496 RTP/AVP 0 101

15:57:15.893 SIP.STACK MSG         a=maxptime:150

15:57:15.893 SIP.STACK MSG         a=sendrecv

15:57:15.893 SIP.STACK MSG         a=rtpmap:0 PCMU/8000

15:57:15.893 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

15:57:15.893 SIP.STACK MSG         a=fmtp:101 0-15

15:57:15.894 SIP.STACK MSG

15:57:17.387 SIP.STACK MSG     Tx: UDP src=XX.XX.XX.186:5060 dst=XX.XXX.XXX.119:5060

15:57:17.387 SIP.STACK MSG         SIP/2.0 503 Service Unavailable

15:57:17.387 SIP.STACK MSG         From: "NAME,MY"<sip:1XXXXXX4155@XX.XXX.XXX.119>;tag=as2092f99b

15:57:17.387 SIP.STACK MSG         To: <sip:1XXXXXX6766@XX.XX.XX.186;user=phone>;tag=525b580-7f000001-13c4-10ed6-6499f565-10ed6

15:57:17.388 SIP.STACK MSG         Call-ID: 60e4bbe0011748054677fbf339e9a872@XX.XXX.XXX.119:5060

15:57:17.388 SIP.STACK MSG         CSeq: 101 INVITE

15:57:17.388 SIP.STACK MSG         Via: SIP/2.0/UDP XX.XXX.XXX.119:5060;rport=5060;branch=z9hG4bK3d7960b9

15:57:17.388 SIP.STACK MSG         Supported: 100rel,replaces

15:57:17.388 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

15:57:17.388 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R13.2.0.E

15:57:17.388 SIP.STACK MSG         Content-Length: 0

15:57:17.388 SIP.STACK MSG

15:57:17.428 SIP.STACK MSG     Rx: UDP src=XX.XXX.XXX.119:5060 dst=XX.XX.XX.186:5060

15:57:17.428 SIP.STACK MSG         ACK sip:1XXXXXX6766@XX.XX.XX.186;user=phone SIP/2.0

15:57:17.428 SIP.STACK MSG         Via: SIP/2.0/UDP XX.XXX.XXX.119:5060;branch=z9hG4bK3d7960b9;rport

15:57:17.428 SIP.STACK MSG         Max-Forwards: 70

15:57:17.428 SIP.STACK MSG         From: "NAME,MY" <sip:1XXXXXX4155@XX.XXX.XXX.119>;tag=as2092f99b

15:57:17.429 SIP.STACK MSG         To: <sip:1XXXXXX6766@XX.XX.XX.186;user=phone>;tag=525b580-7f000001-13c4-10ed6-6499f565-10ed6

15:57:17.429 SIP.STACK MSG         Contact: <sip:1XXXXXX4155@XX.XXX.XXX.119:5060>

15:57:17.429 SIP.STACK MSG         Call-ID: 60e4bbe0011748054677fbf339e9a872@XX.XXX.XXX.119:5060

15:57:17.429 SIP.STACK MSG         CSeq: 101 ACK

15:57:17.429 SIP.STACK MSG         User-Agent: Asterisk PBX (ScopServ)

15:57:17.430 SIP.STACK MSG         Content-Length: 0

15:57:17.430 SIP.STACK MSG

15:57:17.431 SIP.STACK MSG Found existing transaction for the request message.

15:57:17.455 SIP.STACK MSG     Rx: UDP src=10.2.5.145:5060 dst=10.2.0.145:5060

15:57:17.455 SIP.STACK MSG         OPTIONS sip:10.2.0.145 SIP/2.0

15:57:17.455 SIP.STACK MSG         Via: SIP/2.0/UDP 10.2.5.145:5060;rport;branch=z9hG4bKd1dd4d14a0e11b2d2d56531acbf84fb6

15:57:17.455 SIP.STACK MSG         From: <sip:10.2.5.145>;tag=bf32dffc1bc7fe9d

15:57:17.455 SIP.STACK MSG         To: <sip:10.2.0.145>

15:57:17.456 SIP.STACK MSG         Call-ID: 1bfcf66794824ca43989057802212863

15:57:17.456 SIP.STACK MSG         CSeq: 1878346777 OPTIONS

15:57:17.456 SIP.STACK MSG         Contact: <sip:10.2.5.145:5060;transport=udp>

15:57:17.456 SIP.STACK MSG         Max-Forwards: 70

15:57:17.456 SIP.STACK MSG         Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE

15:57:17.456 SIP.STACK MSG         Supported: timer

15:57:17.457 SIP.STACK MSG         User-Agent: IP Office 9.1.8.0 build 172

15:57:17.457 SIP.STACK MSG         Content-Length: 0

15:57:17.457 SIP.STACK MSG

15:57:17.459 SIP.STACK MSG Found existing transaction for the request message.

15:57:17.459 SIP.STACK MSG     Tx: UDP src=10.2.0.145:5060 dst=10.2.5.145:5060

15:57:17.459 SIP.STACK MSG         SIP/2.0 200 OK

15:57:17.459 SIP.STACK MSG         From: <sip:10.2.5.145>;tag=bf32dffc1bc7fe9d

15:57:17.459 SIP.STACK MSG         To: <sip:10.2.0.145>;tag=525b170-7f000001-13c4-10ec8-5040619c-10ec8

15:57:17.460 SIP.STACK MSG         Call-ID: 1bfcf66794824ca43989057802212863

15:57:17.460 SIP.STACK MSG         CSeq: 1878346777 OPTIONS

15:57:17.460 SIP.STACK MSG         Via: SIP/2.0/UDP 10.2.5.145:5060;rport=5060;branch=z9hG4bKd1dd4d14a0e11b2d2d56531acbf84fb6

15:57:17.460 SIP.STACK MSG         Supported: 100rel,replaces

15:57:17.461 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

15:57:17.461 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R13.2.0.E

15:57:17.461 SIP.STACK MSG         Content-Length: 0

15:57:17.461 SIP.STACK MSG

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Honored Contributor
Honored Contributor

Re: SIP to SIP Gateway

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That call was routed to the PRI on trunk T02 and it looks like the PBX rejected it. Try a call with a destination number on trunk T03.

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New Contributor II

Re: SIP to SIP Gateway

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You are correct that the PBX was rejecting the call. I was forwarding the call route on my Avaya IP office to a dead extension that I though was another active one. I am marking this as answered as I contacted ADTRAN support which helped me further. It turns out that what I was trying to accomplish is not possible without an eSBC license. You need a session boarder controller to route sip to sip trunks. The ADTRAN tech was amazed that my sip to sip calls were even going out. The SBC is needed to route RTP traffic properly as well. What I ended up doing is just using PRI, which my PBX is already configured for. Incoming DIDs are now routing to PBX on PRI. I am using "accecpt $ cost 0" on both SIP trunk and PRI trunk as my provider will handle which numbers are allowed to reach my PBX. I am also using reject statements on PRI along with the accept $, allowing me to route individual numbers to the FXS ports for analog extensions.

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