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g-man
New Contributor

Inbound calls not making it to softswitch

Jump to solution

I am hoping this will be my final question on this specific journey. I have a TA900 that is currently serving an older PBX with PRI. I have also been working on installing a soft switch behind the TA900. I am able to place outbound calls from the softswitch but inbound calls are not making it. Initially I was seeing a 404 error on the adtran but after playing with the config some more I lost that as well? Luckily PRI continues to work as expected!

I tried adding another the following for inbound calls with no success. Not sure how to register a trunk from the PBX to Adtran either.

Voice Trunk T03 type SIP

sip-server primary 192.168.33.1

transfer-mode network

grammer from host local

Voice grouped-Trunk PBX

accept 13235556666 cost 0

Current Config

!

!

clock timezone -8

!

ip subnet-zero

ip classless

ip routing

ipv6 unicast-routing

auto-config

!

ip firewall

no ip firewall alg msn

no ip firewall alg mszone

no ip firewall alg h323

!

!

!

!

!

!

!

!

no dot11ap access-point-control

!

interface eth 0/1

description WAN

ip address  76.10.76.10  255.255.255.248

ip address  76.10.76.11  255.255.255.255  secondary

ip access-policy Public

media-gateway ip primary

no shutdown

!

!

interface eth 0/2

description  (LAN)

ip address  192.168.33.1  255.255.255.0

ip access-policy Private

media-gateway ip primary

no awcp

no shutdown

!

!

!

interface gigabit-eth 0/1

  no ip address

  shutdown

!

!

!

!

interface t1 0/1

  shutdown

!

interface t1 0/2

  shutdown

!

interface t1 0/3

  lbo short 15

  tdm-group 1 timeslots 1-24 speed 64

  no shutdown

!

interface t1 0/4

  shutdown

!

!

interface pri 1

  isdn name-delivery proceeding

  connect t1 0/3 tdm-group 1

  digits-transferred 4

  no shutdown

!

!

interface fxs 0/1

  impedance 600r

  no shutdown

!

interface fxs 0/2

  no shutdown

!

interface fxs 0/3

  no shutdown

!

interface fxs 0/4

  no shutdown

!

interface fxs 0/5

  no shutdown

!

interface fxs 0/6

  no shutdown

!

interface fxs 0/7

  no shutdown

!

interface fxs 0/8

  no shutdown

!

interface fxs 0/9

  no shutdown

!

interface fxs 0/10

  no shutdown

!

interface fxs 0/11

  no shutdown

!

interface fxs 0/12

  no shutdown

!

interface fxs 0/13

  no shutdown

!

interface fxs 0/14

  no shutdown

!

interface fxs 0/15

  no shutdown

!

interface fxs 0/16

  no shutdown

!

interface fxs 0/17

  no shutdown

!

interface fxs 0/18

  no shutdown

!

interface fxs 0/19

  no shutdown

!

interface fxs 0/20

  no shutdown

!

interface fxs 0/21

  no shutdown

!

interface fxs 0/22

  no shutdown

!

interface fxs 0/23

  no shutdown

!

interface fxs 0/24

  no shutdown

!

!

isdn-group 1

  connect pri 1

!

!

!

!

!

!

!

!

ip access-list standard allow-all

  remark allow all traffic

  permit any

!

ip access-list standard mgmt-allow-list

  70.11.11.99

!

ip access-list standard sip-allow-list

  permit hostname xx.com

!

!

ip access-list extended WEB-ACL-3

  permit tcp any  any eq https 

  permit tcp any  any eq ssh 

!

ip access-list extended WEB-ACL-4

remark 1:1 NAT 76.10.76.11 > 192.168.33.11

permit ip any  host 76.10.76.11 

!

ip access-list extended WEB-ACL-5

  remark 1:1 NAT 192.168.33.11 > 76.10.76.11

  permit ip host 192.168.33.11  any   

!

!

ip policy-class Private

  nat source list allow-all interface eth 0/1 overload policy Public

  allow list allow-all self

  nat source list WEB-ACL-5 address 76.10.76.11 overload

!

ip policy-class Public

  nat destination list WEB-ACL-4 address 192.168.33.11

  allow list allow-all self

  allow list WEB-ACL-3 self

!

!

!

ip route 0.0.0.0 0.0.0.0 76.10.76.10

!

no tftp server

no tftp server overwrite

no 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 2 long-distance 1-NXX-NXX-XXXX

!

!

!

!

voice codec-list VOICE

  default

  codec g711ulaw

!

voice codec-list FAX

  codec g711ulaw

!

!

!

voice trunk T01 type sip

  description "SIP"

  match dnis "91-NXX-NXX-XXXX" substitute "1-NXX-NXX-XXXX"

  match dnis "9NXX-XXXX" substitute "1-310-NXX-XXXX"

  match dnis "NXX-NXX-XXXX" substitute "1-NXX-NXX-XXXX"

  match dnis "NXX-XXXX" substitute "1-310-NXX-XXXX"

  sip-server primary 88.88.88.88

  registrar primary 88.88.88.88

  domain "76.10.76.10"

  register 100XXXXXXX auth-name "" password ""

  codec-list VOICE both

  authentication username "" password ""

!

voice trunk T02 type isdn

  description "DSX-1"

  resource-selection linear ascending

  connect isdn-group 1

  no early-cut-through

  match dnis "1800XXXXXXX" substitute "13235551212"

  match dnis "1844XXXXXXX" substitute "13235551212"

  rtp delay-mode adaptive

  codec-list VOICE

!

!

voice grouped-trunk SIP

  trunk T01

  accept $ cost 0

!

!

!

!

voice grouped-trunk ISDN

  trunk T02

  accept 1323XXXXXXX cost 0

!

sip access-class ip "sip-allow-list" in

!

line con 0

  no login

!

line telnet 0 4

  login local-userlist

  password password

  shutdown

  ip access-class mgmt-allow-list in

line ssh 0 4

  login local-userlist

  no shutdown

  ip access-class mgmt-allow-list in

!

!

!

!

!

end

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1 Solution

Accepted Solutions
jayh
Honored Contributor
Honored Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

g-man wrote:

I tried adding another the following for inbound calls with no success. Not sure how to register a trunk from the PBX to Adtran either.

Voice Trunk T03 type SIP

sip-server primary 192.168.33.1

transfer-mode network

grammer from host local

Voice grouped-Trunk PBX

accept 13235556666 cost 0

Try adding:

Voice grouped-Trunk PBX

trunk T03

accept 13235556666 cost 0

View solution in original post

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15 Replies
g-man
New Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

I was able to get the log of my calls and it looks like 76.10.76.11 which is the secondary address is not responding to invites. Shouldn't

ip access-list standard sip-allow-list

  permit hostname xx.com

allow the sip traffic to the switch?

jayh
Honored Contributor
Honored Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

g-man wrote:

I tried adding another the following for inbound calls with no success. Not sure how to register a trunk from the PBX to Adtran either.

Voice Trunk T03 type SIP

sip-server primary 192.168.33.1

transfer-mode network

grammer from host local

Voice grouped-Trunk PBX

accept 13235556666 cost 0

Try adding:

Voice grouped-Trunk PBX

trunk T03

accept 13235556666 cost 0

View solution in original post

0 Kudos
jayh
Honored Contributor
Honored Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

g-man wrote:

I was able to get the log of my calls and it looks like 76.10.76.11 which is the secondary address is not responding to invites. Shouldn't

ip access-list standard sip-allow-list

permit hostname xx.com

allow the sip traffic to the switch?

OK, it looks like there are a couple of oddities here.

Your softswitch is on the LAN with an address of 192.168.33.XX, correct?

Add that private IP to your sip-allow-list .

You have trunk T03 pointing to your own interface as the SIP server. Point it to the IP of your softswitch. On the softswitch, point its SIP server address to 192.168.33.1.

Also note that a few firmware revisions back, Adtran deliberately broke the ability of TA900 devices to process some SIP-to-SIP calls unless you purchase an extra SBC license so you may need that to move forward. It's relatively cheap but quite an annoyance.

g-man
New Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

jayh,

Thank you so much for the assistance. I did as you suggested and still nothing. What is odd is that the call is not even making it to the TA. I tried SIP debug but I do not see anything. When I ask the SIP provider they say they are not getting a response to the invite. I would at least expect to see something. I am sending calls to the secondary address on the TA. Would the license be causeing such an isue

g-man
New Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

Is there any way to make this work directly between the softswitch and SIP Provider and bypass the adtran for SIP? I can register and make outbound calls going directly to the provider, just need a way to get the calls to the softswitch. Basically just use it as a Router

g-man
New Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

I inquired with adtran about the license. It seems like something is blocking my invite from the provider yet I have other calls that are processing?

Routing a call from one SIP trunk to another SIP trunk does require an SBC license, however if the ADTRAN receives an INVITE, it will at least acknowledge the packet, as long as the request-URI has the ADTRAN's IP address (addressed to the ADTRAN). If the SIP packet is not addressed to the ADTRAN, then the ADTRAN will not respond.

jayh
Honored Contributor
Honored Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

g-man wrote:

jayh,

Thank you so much for the assistance. I did as you suggested and still nothing. What is odd is that the call is not even making it to the TA. I tried SIP debug but I do not see anything. When I ask the SIP provider they say they are not getting a response to the invite.

Is this the same SIP provider that is sending calls to the PRI, and is the provider sending them in the same manner? Don't use the secondary address, use the main interface address just as you to for calls to the PRI.

What happens if you configure the grouped-trunk on the softswitch T03 to accept one of the numbers now routed to the PRI? Does that call now go to the softswitch?

jayh
Honored Contributor
Honored Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

g-man wrote:

Is there any way to make this work directly between the softswitch and SIP Provider and bypass the adtran for SIP? I can register and make outbound calls going directly to the provider, just need a way to get the calls to the softswitch. Basically just use it as a Router

You could put a switch on the public side ahead of the Adtran and configure the softswitch to be directly on the Internet. Make sure that you have security very well locked down on the softswitch. It also makes calls between the softswitch and the PRI take a sub-optimal path. I'd use the Adtran even if it means shelling out for the license or going back to older firmware before Adtran crippled it. More flexible and secure.

As far as registering the softswitch to the Adtran, you can do this but there is really no need to if it's directly on the LAN connected to the Adtran. Just reference it by IP address in the trunk.

g-man
New Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

I just tried that exact thing and the call does not make it to the softswitch. I don't even see the call hit the adtran. I pointed the softswitch calls to the Public IP of the secondary interface, the PBX is going to the first.

g-man
New Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

I contacted adtran and they said I didn't need the license? I have no problem getting a license. The thing that puzzles me is why am I not seeing the invite on calls routed to the softswitch?

g-man
New Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

I just got an invite from the PRI side.

09:40:42.181 SIP.STACK MSG     Rx: UDP src=88.88.88.88:5060 dst=76.10.76.11:5060
09:40:42.181 SIP.STACK MSG         INVITE sip:13237360311@76.10.76.11 SIP/2.0
09:40:42.182 SIP.STACK MSG         Via: SIP/2.0/UDP 88.88.88.88:5060;branch=z9hG4bKcvpnng008g81vf0fn7k0.1
09:40:42.182 SIP.STACK MSG         From: "Bernard,St." <sip:+16264919734@88.88.88.88>;tag=SDphks201-gK0458d68d
09:40:42.182 SIP.STACK MSG         To: <sip:13237360311@76.10.76.11>
09:40:42.182 SIP.STACK MSG         Remote-Party-ID: "Bernard,St." <sip:+16264919734@88.88.88.88:5060>;privacy=off;screen=yes
09:40:42.182 SIP.STACK MSG         Call-ID: SDphks201-7282d27c2edb07ed73bf4900e7e32ad7-523g533
09:40:42.182 SIP.STACK MSG         CSeq: 984876 INVITE
09:40:42.182 SIP.STACK MSG         Max-Forwards: 68
09:40:42.183 SIP.STACK MSG         Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
09:40:42.183 SIP.STACK MSG         Accept: application/sdp
09:40:42.183 SIP.STACK MSG         Contact: <sip:88.88.88.88:5060;did=72b.0dc32b76;transport=udp>
09:40:42.183 SIP.STACK MSG         Supported: replaces
09:40:42.183 SIP.STACK MSG         Content-Length: 254
09:40:42.183 SIP.STACK MSG         Content-Disposition: session; handling=required
09:40:42.184 SIP.STACK MSG         Content-Type: application/sdp
09:40:42.184 SIP.STACK MSG
09:40:42.184 SIP.STACK MSG         v=0
09:40:42.184 SIP.STACK MSG         o=- 3788767953 759660 IN IP4 99.99.99.99
09:40:42.184 SIP.STACK MSG         s=-
09:40:42.184 SIP.STACK MSG         c=IN IP4 99.99.99.99
09:40:42.184 SIP.STACK MSG         t=0 0
09:40:42.185 SIP.STACK MSG         m=audio 4666 RTP/AVP 0 18 101
09:40:42.185 SIP.STACK MSG         a=rtpmap:0 PCMU/8000
09:40:42.185 SIP.STACK MSG         a=rtpmap:18 G729/8000
09:40:42.185 SIP.STACK MSG         a=fmtp:18 annexb=no
09:40:42.185 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000
09:40:42.185 SIP.STACK MSG         a=fmtp:101 0-15
09:40:42.185 SIP.STACK MSG         a=sendrecv
09:40:42.186 SIP.STACK MSG         a=ptime:20
09:40:42.186 SIP.STACK MSG
09:40:42.190 SIP.STACK MSG     Tx: UDP src=76.10.76.11:5060 dst=88.88.88.88:5060
09:40:42.190 SIP.STACK MSG         SIP/2.0 100 Trying
09:40:42.190 SIP.STACK MSG         From: "Bernard,St."<sip:+16264919734@88.88.88.88>;tag=SDphks201-gK0458d68d
09:40:42.190 SIP.STACK MSG         To: <sip:13237360311@76.10.76.11>
09:40:42.190 SIP.STACK MSG         Call-ID: SDphks201-7282d27c2edb07ed73bf4900e7e32ad7-523g533
09:40:42.190 SIP.STACK MSG         CSeq: 984876 INVITE
09:40:42.191 SIP.STACK MSG         Via: SIP/2.0/UDP 88.88.88.88:5060;branch=z9hG4bKcvpnng008g81vf0fn7k0.1
09:40:42.191 SIP.STACK MSG         Contact: <sip:13237360311@76.10.76.11:5060;transport=UDP>
09:40:42.191 SIP.STACK MSG         Supported: 100rel,replaces
09:40:42.191 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
09:40:42.191 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_900e_3rd_Gen/R10.9.5.E
09:40:42.191 SIP.STACK MSG         Content-Length: 0
09:40:42.192 SIP.STACK MSG
09:40:42.197 SIP.STACK MSG     Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
09:40:42.197 SIP.STACK MSG         INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0
09:40:42.198 SIP.STACK MSG         From: "Bernard,St." <sip:6264919734@76.10.76.11:5060;transport=UDP>;tag=63139890-a0a5801-13c4-1401a-2ea8031f-1401a
09:40:42.198 SIP.STACK MSG         To: <sip:13237360311@192.168.33.3:5060>
09:40:42.198 SIP.STACK MSG         Call-ID: 63161588-a0a5801-13c4-1401a-113af5b4-1401a@76.10.76.11
09:40:42.198 SIP.STACK MSG         CSeq: 1 INVITE
09:40:42.198 SIP.STACK MSG         Via: SIP/2.0/UDP 76.10.76.11:5060;branch=z9hG4bK-1401a-4e266b6-3abb03d
09:40:42.198 SIP.STACK MSG         Max-Forwards: 70
09:40:42.199 SIP.STACK MSG         Supported: 100rel,replaces
09:40:42.199 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
09:40:42.199 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_900e_3rd_Gen/R10.9.5.E
09:40:42.199 SIP.STACK MSG         Contact: <sip:6264919734@76.10.76.11:5060;transport=UDP>
09:40:42.199 SIP.STACK MSG         Content-Type: application/sdp
09:40:42.199 SIP.STACK MSG         Content-Length: 234
09:40:42.199 SIP.STACK MSG
09:40:42.200 SIP.STACK MSG         v=0
09:40:42.200 SIP.STACK MSG         o=- 3788767953 759660 IN IP4 99.99.99.99
09:40:42.200 SIP.STACK MSG         s=-
09:40:42.200 SIP.STACK MSG         c=IN IP4 99.99.99.99
09:40:42.200 SIP.STACK MSG         t=0 0
09:40:42.200 SIP.STACK MSG         m=audio 4666 RTP/AVP 0 101
09:40:42.200 SIP.STACK MSG         a=sendrecv
09:40:42.201 SIP.STACK MSG         a=ptime:20
09:40:42.201 SIP.STACK MSG         a=rtpmap:0 PCMU/8000
09:40:42.201 SIP.STACK MSG         a=silenceSupp:off - - - -
09:40:42.201 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000
09:40:42.201 SIP.STACK MSG         a=fmtp:101 0-15
09:40:42.201 SIP.STACK MSG
09:40:42.702 SIP.STACK MSG     Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
09:40:42.702 SIP.STACK MSG         INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0
09:40:42.702 SIP.STACK MSG         From: "Bernard,St." <sip:6264919734@76.10.76.11:5060;transport=UDP>;tag=63139890-a0a5801-13c4-1401a-2ea8031f-1401a
09:40:42.702 SIP.STACK MSG         To: <sip:13237360311@192.168.33.3:5060>
09:40:42.702 SIP.STACK MSG         Call-ID: 63161588-a0a5801-13c4-1401a-113af5b4-1401a@76.10.76.11
09:40:42.703 SIP.STACK MSG         CSeq: 1 INVITE
09:40:42.703 SIP.STACK MSG         Via: SIP/2.0/UDP 76.10.76.11:5060;branch=z9hG4bK-1401a-4e266b6-3abb03d
09:40:42.703 SIP.STACK MSG         Max-Forwards: 70
09:40:42.703 SIP.STACK MSG         Supported: 100rel,replaces
09:40:42.703 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
09:40:42.703 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_900e_3rd_Gen/R10.9.5.E
09:40:42.704 SIP.STACK MSG         Contact: <sip:6264919734@76.10.76.11:5060;transport=UDP>
09:40:42.704 SIP.STACK MSG         Content-Type: application/sdp
09:40:42.704 SIP.STACK MSG         Content-Length: 234
09:40:42.704 SIP.STACK MSG
09:40:42.704 SIP.STACK MSG         v=0
09:40:42.704 SIP.STACK MSG         o=- 3788767953 759660 IN IP4 99.99.99.99
09:40:42.705 SIP.STACK MSG         s=-
09:40:42.705 SIP.STACK MSG         c=IN IP4 99.99.99.99
09:40:42.705 SIP.STACK MSG         t=0 0
09:40:42.705 SIP.STACK MSG         m=audio 4666 RTP/AVP 0 101
09:40:42.705 SIP.STACK MSG         a=sendrecv
09:40:42.705 SIP.STACK MSG         a=ptime:20
09:40:42.705 SIP.STACK MSG         a=rtpmap:0 PCMU/8000
09:40:42.706 SIP.STACK MSG         a=silenceSupp:off - - - -
09:40:42.706 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000
09:40:42.706 SIP.STACK MSG         a=fmtp:101 0-15
09:40:42.706 SIP.STACK MSG
09:40:43.707 SIP.STACK MSG     Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
09:40:43.707 SIP.STACK MSG         INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0
09:40:43.707 SIP.STACK MSG         From: "Bernard,St." <sip:6264919734@76.10.76.11:5060;transport=UDP>;tag=63139890-a0a5801-13c4-1401a-2ea8031f-1401a
09:40:43.707 SIP.STACK MSG         To: <sip:13237360311@192.168.33.3:5060>
09:40:43.707 SIP.STACK MSG         Call-ID: 63161588-a0a5801-13c4-1401a-113af5b4-1401a@76.10.76.11
09:40:43.708 SIP.STACK MSG         CSeq: 1 INVITE
09:40:43.708 SIP.STACK MSG         Via: SIP/2.0/UDP 76.10.76.11:5060;branch=z9hG4bK-1401a-4e266b6-3abb03d
09:40:43.708 SIP.STACK MSG         Max-Forwards: 70
09:40:43.708 SIP.STACK MSG         Supported: 100rel,replaces
09:40:43.708 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
09:40:43.708 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_900e_3rd_Gen/R10.9.5.E
09:40:43.709 SIP.STACK MSG         Contact: <sip:6264919734@76.10.76.11:5060;transport=UDP>
09:40:43.709 SIP.STACK MSG         Content-Type: application/sdp
09:40:43.709 SIP.STACK MSG         Content-Length: 234
09:40:43.709 SIP.STACK MSG
09:40:43.709 SIP.STACK MSG         v=0
09:40:43.709 SIP.STACK MSG         o=- 3788767953 759660 IN IP4 99.99.99.99
09:40:43.709 SIP.STACK MSG         s=-
09:40:43.710 SIP.STACK MSG         c=IN IP4 99.99.99.99
09:40:43.710 SIP.STACK MSG         t=0 0
09:40:43.710 SIP.STACK MSG         m=audio 4666 RTP/AVP 0 101
09:40:43.710 SIP.STACK MSG         a=sendrecv
09:40:43.710 SIP.STACK MSG         a=ptime:20
09:40:43.710 SIP.STACK MSG         a=rtpmap:0 PCMU/8000
09:40:43.711 SIP.STACK MSG         a=silenceSupp:off - - - -
09:40:43.711 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000
09:40:43.711 SIP.STACK MSG         a=fmtp:101 0-15
09:40:43.711 SIP.STACK MSG

09:40:44.212 SIP.STACK MSG
09:40:45.205 SIP.STACK MSG     Tx: UDP src=76.10.76.11:5060 dst=88.88.88.88:5060
09:40:45.205 SIP.STACK MSG         SIP/2.0 503 Service Unavailable
09:40:45.205 SIP.STACK MSG         From: "Bernard,St."<sip:+16264919734@88.88.88.88>;tag=SDphks201-gK0458d68d
09:40:45.205 SIP.STACK MSG         To: <sip:13237360311@76.10.76.11>;tag=63138c18-a0a5801-13c4-1401d-4ad6b38d-1401d
09:40:45.206 SIP.STACK MSG         Call-ID: SDphks201-7282d27c2edb07ed73bf4900e7e32ad7-523g533
09:40:45.206 SIP.STACK MSG         CSeq: 984876 INVITE
09:40:45.206 SIP.STACK MSG         Via: SIP/2.0/UDP 88.88.88.88:5060;branch=z9hG4bKcvpnng008g81vf0fn7k0.1
09:40:45.206 SIP.STACK MSG         Supported: 100rel,replaces
09:40:45.206 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
09:40:45.206 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_900e_3rd_Gen/R10.9.5.E
09:40:45.206 SIP.STACK MSG         Content-Length: 0

g-man
New Contributor

Re: Inbound calls not making it to softswitch

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Voice Verbose

09:54:32.108 TM.T01 01 SipTM_Idle           rcvd SIP call-leg request: INVITE

09:54:32.108 TM.T01 01 SipTM_Idle           call-leg -> Offering

09:54:32.109 TM.T01 01 SipTM_Idle           State change      >> SipTM_Idle->SipTM_Trying

09:54:32.109 TM.T01 01 SipTM_Trying         SDP offer is not loopback request

09:54:32.109 TM.T01 01 SipTM_Trying         Processing From for Caller-ID.

09:54:32.110 TM.T01 01 T01 01 SipTM_Trying         e164 calling number converted to dialstring 13235551212

09:54:32.110 TM.T01 01 SipTM_Trying         Caller ID Name   = "Bernard, St"

09:54:32.110 TM.T01 01 SipTM_Trying         Caller ID Number = "13235551212"

09:54:32.110 TM.T01 01 SipTM_Trying         info: unable to set redirect number(s) from INVITE

09:54:32.110 TM.T01 01 SipTM_Trying         sent: TA->InboundCall

09:54:32.111 TM.T01 01 Looking up source address for destination 69.69.69.69

09:54:32.111 TM.T01 01 call-leg (0x0x63161390) -> src: 76.10.76.10: 5060  dst: 69.69.69.69 : 5060

09:54:32.112 TM.T01 01 SipTM_Trying         sent: 100 Trying

09:54:32.112 TA.T01 01 TAIdle               rcvd: inboundCall from TM

09:54:32.113 TA.T01 01 State change      >> TAIdle->TAInboundCall (TAS_Calling)

09:54:32.113 TA.T01 01 Failed - DID translation: no match for 13237360311, using 13237360311

09:54:32.113 TA.T01 01 TAIdle               sent: call to SB

09:54:32.113 TM.T01 01 SipTM_Trying         tachg -> TAInboundCall

09:54:32.113 TM.T01 01 SipTM_Trying         State change      >> SipTM_Trying->SipTM_Pending

09:54:32.114 SB.CALL 479 Idle                 Called the call routine with 13237360311

09:54:32 SB.TGMgr For dialed number 13237360311, against template $, on TrunkGroup SIP, the score is 500

09:54:32 SB.TGMgr For dialed number 13237360311, against template 13237360311, on TrunkGroup PBX, the score is 12000

09:54:32.114 SB.CCM isMappable:

09:54:32.115 SB.CCM  :  Call Struct 0x0x7520e610 :   Call-ID = 479

09:54:32.115 SB.CCM  :  Org Acct = T01    Dst Acct = T03

09:54:32.115 SB.CCM  :  Org Port ID = SipTrunk 0/0   Dst Port ID = unknown 0/0

09:54:32.115 SB.CCM  :  SDP Transaction = CallID: 479

09:54:32.115 SB.CCM  :  SDP Offer = 0x75201310, (88.88.88.88:44666)

09:54:32.116 SB.CCM isMappable: Call Connection Type is RTP_TO_RTP

09:54:32.116 SB.CCM handleRtpToRtp: Modifying SDP Offer

09:54:32.116 SB.CCM translateOffer: offer codec list: PCMU G729

09:54:32.117 SB.CCM translateOffer: revised offer codec list: PCMU

09:54:32.117 SB.CCM translateOffer: codec list after answerer: PCMU

09:54:32.118 SB.CCM translateOffer: DTMF signaling: answerer has no restrictions configured, passing offer(NTE 101) through

09:54:32.118 SB.CCM translateOffer: success

09:54:32.118 MEDIA.MANAGER Allocating media port.

09:54:32.119 MEDIA.MANAGER getSubstitutePort: No matching callIdMap entry found for call 479

09:54:32.119 MEDIA.MANAGER Call ID map : Added new entry : call ID 479 : session -3249275585INIP488.88.88.88 : version 951176 : index 1712

09:54:32.119 MEDIA.MANAGER New media entry : type(0), callID(479), sessionID(-3249275585INIP488.88.88.88), original IP(88.88.88.88) ports(44666-44667), substitute IP(::) ports(11712-11713), RtpChannel(NULL), connection(0x0x75313710), sdpOverride(0), me(0x0x75202310). No RtpChannel

09:54:32.119 SB.CALL 479 Idle                 Call sent from T01 to T03 (13237360311)

09:54:32.120 SB.CALL 479 State change      >> Idle->Delivering

09:54:32.120 TA.T01 01 TAInboundCall        CallResp event accepted

09:54:32.120 TA.T01 01 State change      >> TAInboundCall->TAConnectWaitIn (TAS_Calling)

09:54:32.120 TA.T03 91 State change      >> TAIdle->TAOutGoing (TAS_Delivering)

09:54:32.121 TM.T03 91 SipTM_Idle           State change      >> SipTM_Idle->Delivering

09:54:32.121 TM.T03 91 Delivering           Applying E.164 settings to called party number (13237360311)

09:54:32.121 TM.T03 91 Delivering           Skipping E.164 conversion due to voice international-prefix setting

09:54:32.121 TM.T03 91 Delivering           Applying E.164 settings to calling party number (13235551212)

09:54:32.121 TM.T03 91 Delivering           From user grammar setting is: domestic

09:54:32.122 TM.T03 91 Delivering           Skipping E.164 conversion due to From user grammar setting

09:54:32.122 TM.T03 91 Looking up source address for destination 192.168.33.1

09:54:32.122 TM.T03 91 call-leg (0x0x63161f60) -> src: 76.10.76.10: 5060  dst: 192.168.33.1 : 5060

09:54:32.123 TM.T03 91 SDP DPI call ID 479 : No media bin.

09:54:32.123 TM.T03 91 Processing new SDP entries.

09:54:32.123 TM.T03 91 Checking for internal Media Gateway IP Address

09:54:32.123 TM.T03 91 RTP Channel is NULL, Media Gateway must not be involved in call

09:54:32.124 TM.T03 91 Undo of previous operation not required (RTP NAT Entry for 88.88.88.88:44666 not found)

09:54:32.124 TM.T03 91 Checking for internal Media Gateway IP Address

09:54:32.124 TM.T03 91 Given RTP Channel is null, checking for hairpinned RTP Channel

09:54:32.124 TM.T03 91 RTP Channel is NULL, Media Gateway must not be involved in call

09:54:32.124 TM.T03 91 Checking need for firewall traversal

09:54:32.124 TM.T03 91 Testing firewall policies

09:54:32.125 TM.T03 91 NAT not required, no need for firewall traversal here

09:54:32.126 TM.T03 91 Delivering           call-leg -> Inviting

09:54:32.127 TM.T03 91 Delivering           sent: INVITE

09:54:32.127 SB.CALL 479 Delivering           Called the deliverResponse routine from Delivering

09:54:32.127 SB.CALL 479 Delivering           DeliverResponse(accept) sent from T03 to T01

09:54:32.128 TA.T01 01 TAConnectWaitIn      deliverResponse event accepted

09:54:32.128 TA.T01 01 TAConnectWaitIn      ERROR! deliverResponse ignored

09:54:32 SB.CallStructObserver 479 Created

09:54:32 SB.CallStructObserver 479 <-> SDvpmc201-8d7cb69aee75599fcfc00a6d83e883a0-523g533

09:54:35.128 TM.T03 91 INVITE rollover timeout

09:54:35.128 TM.T03 91 Delivering           Sip_CreateCallLegNextServer with default validator

09:54:35.128 TM.T03 91 Delivering           State change      >> Delivering->SipTM_Closing

09:54:35.129 TM.T03 91 SipTM_Closing        sent: TA->Clear

09:54:35.129 TM.T03 91 SipTM_Closing        call-leg -> Terminated

09:54:35.129 TA.T03 91 TAOutGoing           rcvd: clear from TM

09:54:35.129 TA.T03 91 State change      >> TAOutGoing->TATrunkClearing (TAS_Clearing)

09:54:35.130 TM.T03 91 SipTM_Closing        tachg -> TATrunkClearing

09:54:35.130 TM.T03 91 SipTM_Closing        State change      >> SipTM_Closing->SipTM_Terminated

09:54:35.130 TM.T03 91 SipTM_Terminated     sent: TA->AppearanceOff

09:54:35.130 TM.T03 91 SipTM_Terminated     State change      >> SipTM_Terminated->SipTM_Idle

09:54:35.130 SB.CALL 479 Delivering           Called the clearCall routine

09:54:35.131 SB.CALL 479 Delivering           SIP Proxy rejected call to 13237360311 for survivability - no matching Proxy user

09:54:35.131 SB.CALL 479 Delivering           No available resources on call from T01 to T03 (last attempt)

09:54:35.131 SB.CALL 479 State change      >> Delivering->Clearing

09:54:35.131 TA.T03 91 TATrunkClearing      rcvd: appearance off from TM

09:54:35.131 TA.T03 91 State change      >> TATrunkClearing->TAClearingComplete (TAS_Clearing)

09:54:35.132 TA.T03 91 TATrunkClearing      Processing an appearance OFF

09:54:35.132 TA.T01 01 TAConnectWaitIn      ClearCall event accepted

09:54:35.132 TA.T01 01 State change      >> TAConnectWaitIn->TAClearingComplete (TAS_Clearing)

09:54:35.132 TM.T01 01 SipTM_Pending        tachg -> TAClearingComplete

09:54:35.132 TM.T01 01 SipTM_Pending        State change      >> SipTM_Pending->SipTM_CallFail

09:54:35.133 TM.T01 01 SipTM_CallFail       call-leg -> Disconnected

09:54:35.134 TM.T01 01 SipTM_CallFail       CallLegStateChanged to Disconnected - TM change to closing state.

09:54:35.134 TM.T01 01 SipTM_CallFail       State change      >> SipTM_CallFail->SipTM_Closing

09:54:35.134 TM.T01 01 SipTM_Closing        sent: TA->Clear

09:54:35.134 TM.T01 01 SipTM_CallFail       sent: 503

09:54:35.134 TM.T01 01 SipTM_Closing        State change      >> SipTM_Closing->SipTM_Terminated

09:54:35.135 TM.T01 01 SipTM_Terminated     sent: TA->AppearanceOff

09:54:35.135 TM.T01 01 SipTM_Terminated     State change      >> SipTM_Terminated->SipTM_Idle

09:54:35.135 SB.CALL 479 Clearing             Called the clearResponse routine

09:54:35.135 SB.CALL 479 State change      >> Clearing->CallIdlePending

09:54:35.136 SB.CCM release:

09:54:35.136 SB.CCM  :  Call Struct 0x0x7520e610 :   Call-ID = 479

09:54:35.136 SB.CCM  :  Org Acct = T01    Dst Acct = T03

09:54:35.136 SB.CCM  :  Org Port ID = SipTrunk 0/0   Dst Port ID = SipTrunk 0/0.290

09:54:35.136 SB.CCM  :  SDP Transaction = CallID: 479

09:54:35.137 SB.CCM  :  SDP Offer = 0x75201310, (88.88.88.88:44666)

09:54:35.137 SB.CCM release: Call Connection Type is RTP_TO_RTP

09:54:35.137 SB.CALL 479 CallIdlePending      ClearResponse sent from T01 to T03

09:54:35.137 TA.T01 01 TAClearingComplete   rcvd: clear from TM

09:54:35.137 TA.T01 01 TAClearingComplete   rcvd: appearance off from TM

09:54:35.137 TA.T01 01 TAClearingComplete   Clear Local Variables

09:54:35.138 TA.T01 01 State change      >> TAClearingComplete->TAIdle (TAS_Idle)

09:54:35.138 TM.T01 01 SipTM_Idle           tachg -> TAIdle

09:54:35.138 TA.T03 91 TAClearingComplete   clearResponse event accepted

09:54:35.138 TA.T03 91 TAClearingComplete   Clear Local Variables

09:54:35.138 TA.T03 91 State change      >> TAClearingComplete->TAIdle (TAS_Idle)

09:54:35.139 TM.T03 91 SipTM_Idle           tachg -> TAIdle

09:54:35 SB.CallStructObserver 479 Finalized

2019.04.11 09:54:36 SMDR 479        04/11/2019 09:54:32      0.0 0    E  00/00 Bernard, St 13235551212      00/00 T03             13237360311     0 N  no debug voice verbose

RDLINC#

jayh
Honored Contributor
Honored Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

Your softswitch on 192.168.33.3 isn't responding to the invite, or its response is being filtered.

From your log the TA900 sent three invites and received no response.

09:40:42.197 SIP.STACK MSG Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
09:40:42.197 SIP.STACK MSG    

INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0

09:40:42.702 SIP.STACK MSG Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
09:40:42.702 SIP.STACK MSG     INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0

09:40:43.707 SIP.STACK MSG Tx: UDP src=76.10.76.11:5060 dst=192.168.33.3:5060
09:40:43.707 SIP.STACK MSG     INVITE sip:13237360311@192.168.33.3:5060 SIP/2.0

After the third attempt the Adtran gave up and sent 503 to your SIP provider.

09:40:45.205 SIP.STACK MSG Tx: UDP src=76.10.76.11:5060 dst=88.88.88.88:5060
09:40:45.205 SIP.STACK MSG    

SIP/2.0 503 Service Unavailable

Is either 192.168.33.0 0.0.0.255 or host 192.168.33.3 in your sip-allow-list? It should be.

Is the softswitch programmed with its SIP server as 192.168.1.1 ? It should be.

Do you have "media-gateway ip primary" set on your LAN interface of 192.168.33.1? You should.

jayh
Honored Contributor
Honored Contributor

Re: Inbound calls not making it to softswitch

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g-man wrote:

I contacted adtran and they said I didn't need the license? I have no problem getting a license. The thing that puzzles me is why am I not seeing the invite on calls routed to the softswitch?

According to the log snippet, the call is being sent to the softswitch, but the TA900 isn't getting a response or the response is being filtered.

g-man
New Contributor

Re: Inbound calls not making it to softswitch

Jump to solution

I got it working!!! I still cannot route calls via the secondary address on eth0/1 but it will have to do.  I was on a deadline to have this working by tomorrow and it looks like I am going to make it. Thank you sooo much.