HI
i have a adtran 980e that is working fine on my opensips right now i am trying to move it to an freeswitch and its not working
the tech guy from freeswitch told me that its an issue with the from header see below
This is the BAD FROM Header the device is sending
2018/08/02 18:16:20.407581 96.xxx.xxx.211:5060 -> 165.xxx.xxx.243:5060
INVITE sip:1nxxnxxxxxxx@sbc.abc.cloud:5060 SIP/2.0
From: <sip:sbc.abc.cloud:5060;transport=UDP>;tag=4de79a0-7f000001-13c4-3a7d-70f521b2-3a7d
To: <sip:nxxnxxxxxxx@sbc.abc.cloud:5060>
Call-ID: 4e29660-7f000001-13c4-3a7d-2854f0b-3a7d@sbc.dataphone.cloud
CSeq: 1 INVITE
Via: SIP/2.0/UDP 96.57.140.211:5060;branch=z9hG4bK-3a7d-e4797f-4cee2846
Max-Forwards: 70
Supported: 100rel,replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R11.10.1.E
Contact: <sip:96.xxx.xxx.xxx:5060;transport=UDP>
Content-Type: application/sdp
Content-Length: 257
This is a good one
2018/08/02 18:18:56.053365 54.xxx.xxx.163:5090 -> 165.xxx.xxx.243:5060
INVITE sip:18454263110@96.xxx.xxx.211:5060 SIP/2.0
Via: SIP/2.0/UDP 54.xxx.xxx.163:5090;rport;branch=z9hG4bKpN30QNNX8By4c
Max-Forwards: 68
From: "7185695157" <sip:7185695157@54.xxx.xxx.163>;tag=F2QgjBNZ1F4Na
To: <sip:18454263110@96.xxx.xxx.211:5060>
Call-ID: 715c3bf4-1123-1237-8689-005056976403
CSeq: 126280921 INVITE
Contact: <sip:mod_sofia@54.xxx.xxx.163:5090>
User-Agent: cgrtbilling_V2.5
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 294
X-cid: 1282146318_129495037@208.xxx.xxx.203
X-FS-Support: update_display,send_info
Remote-Party-ID: "7185695157" <sip:7185695157@54.xxx.xxxx.163>;party=calling;screen=yes;privacy=off
BTW the PBX is not sending any callerID info so i am sending it from the adtran by using the ANI Substitution
ok, once you get the debug i can see exactly what is going on, we can make the changes necessary to make it work.
The ANI isn't appearing in the INVITE. Rather than using ANI substitution, try the following command on the trunk incoming from the PBX:
caller-id-override number-inbound 7185695157
This is for outgoing calls do I still write inbound
mtr wrote:
This is for outgoing calls do I still write inbound
Yes. The commands are looking from the standpoint of the interface or trunk as seen by the Adtran box. The caller-id is inbound to the Adtran from the PBX.
ok i will report back tomorrow on this
still not working
this is how my config looks like
voice trunk T01 type sip
description "ipsbc"
caller-id-override number-inbound 845XXXXX10
match dnis "NXX-XXXX" substitute "1845-NXX-XXXX"
match dnis "NXX-NXX-XXXX" substitute "1NXX-NXX-XXXX"
sip-server primary sip.abc.cloud
outbound-proxy primary sip.abc.cloud
dial-string source to
grammar from user international
Moshe,
The ANI substitution command should work and you shouldn't need the caller-id command.
What I need is your full config. please post that and remove passwords.
next enter the following debug during an outbound call and attach it in a .txt file.
debug sip stack messages
debug sip cldu
debug voice verbose
-Mark
!
!
! ADTRAN, Inc. OS version R11.10.1.E
! Boot ROM version 14.05.00.SA
! Platform: Total Access 908e (2nd Gen), part number 4242908L1
! Serial number CFG1072894
!
!
hostname "TA908e"
enable password xxxxx
!
!
clock timezone -5-Eastern-Time
!
ip subnet-zero
ip classless
ip default-gateway xxx.xxx.xxx.xxx
ip routing
ipv6 unicast-routing
!
!
name-server 8.8.8.8
!
!
no auto-config
!
event-history on
no logging forwarding
no logging email
!
no service password-encryption
!
username "xxxxx" password "xxxxx"
!
!
ip firewall
ip firewall stealth
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 192.168.1.200 255.255.255.0
media-gateway ip primary
no awcp
no shutdown
!
!
interface eth 0/2
description WAN
ip address xxx.xxx.xxx.xxx 255.255.255.xxx
ip mtu 1500
ip access-policy Public
media-gateway ip primary
no awcp
no shutdown
!
!
!
!
interface t1 0/1
description PRI to PBX 2
tdm-group 2 timeslots 1-24 speed 64
no shutdown
!
interface t1 0/2
no shutdown
!
interface t1 0/3
no shutdown
!
interface t1 0/4
tdm-group 1 timeslots 1-24 speed 64
no shutdown
!
!
interface pri 1
description pri 1
isdn name-delivery setup
connect t1 0/4 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
!
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 fxo 0/0
no shutdown
!
!
isdn-group 1
connect pri 1
!
!
!
!
!
!
!
ip access-list extended Admin
remark Admin Access
permit tcp any any eq ssh log
permit tcp any any eq https log
!
ip access-list extended SIP
remark SIP Service Provider
permit udp host xxx.xxx.xxx.110 any eq 5060
permit udp host xxx.xxx.xxx.243 any eq 5060
!
ip access-list extended T
!
!
!
!
ip policy-class Public
allow list Admin self
allow list SIP self
!
!
!
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-XXX-XXXX
voice dial-plan 2 long-distance 1-NXX-XXX-XXXX
voice dial-plan 3 toll-free MXXX
!
!
!
!
!
!
voice trunk T01 type sip
description "ipsbc"
match dnis "NXX-XXXX" substitute "1845-NXX-XXXX"
match dnis "NXX-NXX-XXXX" substitute "1NXX-NXX-XXXX"
match ani "$" substitute "845426xxxx"
sip-server primary sip.abc.cloud
outbound-proxy primary sip.abc.cloud
dial-string source to
grammar from user international
!
voice trunk T02 type isdn
resource-selection circular descending
connect isdn-group 1
no early-cut-through
t38
rtp delay-mode adaptive
!
!
voice grouped-trunk PRI
trunk T02
accept $ cost 0
!
!
voice grouped-trunk SIP
trunk T01
accept $ cost 0
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
sip grammar request-uri host domain
sip grammar to host domain
!
!
!
!
!
!
!
line con 0
login
!
line telnet 0 4
login
no shutdown
line ssh 0 4
login local-userlist
no shutdown
!
!
!
!
!
end
i will wait for call to start and will post the debug output
also if your using FQDN on your sip server enter the following:
voip name-service host sip.abc.cloud sip udp
voip name-service verification attempts 5 interval 30
then you can see what IPs are resolved in your FQDN:
show voip name-service cache
see if it matches your access list:
ip access-list extended SIP
remark SIP Service Provider
permit udp host xxx.xxx.xxx.110 any eq 5060
permit udp host xxx.xxx.xxx.243 any eq 5060
also i would remove this from T01, not needed
then add your domain to T01
voice trunk t01
no outbound-proxy primary sip.abc.cloud
domain sip.abc.cloud
what are you trying to accomplish with your ani substitution command?
That might be one of your problems.
you probably will need to swap it but will see what the debug says when you place a call.
My PBX is not sending callerID Info i am trying to set the callerID on the adtran Device
Have you tried adding the commands below to you PRI interface?
interface pri 1
calling-party override if-no-CID
calling-party number NPANXXXXXX
ok, once you get the debug i can see exactly what is going on, we can make the changes necessary to make it work.
i did
also i would remove this from T01, not needed
then add your domain to T01
voice trunk t01
no outbound-proxy primary sip.abc.cloud
domain sip.abc.cloud
the and the problem was solved
Moshe,
Thanks for the update on your problem. Glad everything is working now!
-Mark