i gave spend many hours to try yo figure this out
i have an opensips server that i use to sell SIP TRUNK right now i have some Asterisk, freeswitch, Cisco ATA devices that are connected to this switch. i got a customer that has a panasonic PBX with a T1 interface so i went ahead and bought the 908e 2nd generation i spend many hour configuring it i couldnt get it to work it was sending the wrong info so i was not able to Auth the outbound calls
finally i got it to register and right now i am getting server 501 on my opensips this is the error from my logfile
/usr/sbin/opensips[24112]: ERROR:core:parse_sip_msg_uri: bad uri <sip:3476783993@68.68.120.146:5060:5060>
so i went ahead and removed all the 5060 that i saw in the web gui and i am still seeing the error line with the 5060 twice
this device is sitting on the local network behind a router
this is my config file
!
!
hostname "TA908e"
enable password xxxxxxxx
!
!
clock timezone -5-Eastern-Time
!
ip subnet-zero
ip classless
ip default-gateway 192.168.2.1
ip routing
ipv6 unicast-routing
!
!
name-server 8.8.8.8 192.168.2.249
!
!
no auto-config
!
event-history on
no logging forwarding
no logging email
!
no service password-encryption
!
username "admin" password "xxxxxxxxx"
!
!
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 192.168.2.60 255.255.255.0
no awcp
no shutdown
media-gateway ip primary
!
!
interface eth 0/2
no ip address
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
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
!
!
!
!
!
!
!
!
!
!
!
ssh-server 223
no tftp server
no tftp server overwrite
http server 82
http secure-server 445
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 trunk T01 type sip
Description "DPhone"
sip-server primary sip.xxxxx.cloud
domain "108.58.25.30"
sip-keep-alive options 30
codec-list Trunks both
no diversion-supported
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
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
line con 0
login
!
line telnet 0 4
login
no shutdown
line ssh 0 4
login local-userlist
no shutdown
!
!
!
!
!
end
Option 501 errors are not always error messages. They are a keep alive message that keeps the ports open and registration established. If you are seeing the messages but calls are woring normally the message can be ignored. If calls are not working normally then we will need to see the output of debug sip stack messages and debug voice verbose from the Adtran during a call. Also I see where you have the Domain command on the SIP Trunk but no Grammer Statements telling the how to use it, so if the Adtran is not able to register or make calls you may want to try adding the following to the config:
ip sip grammar from host domain
ip sip grammar to host domain
If your Softswitch requires registration you have to add it to your trunk as well, if using IP based trust then the current config should be fine. Add the following to your Voice Trunk T01 to add registration:
registrar primary sip.xxxxx.cloud
register UsernameHere auth-name "UsernameHere" password "PasswordHere"
authentication username "UsernameHere" password "PasswordHere"
John Wable
I would do "debug sip stack messages" using the Adtran CLI and see if the URI sent from the Adtran does indeed list the port twice. Based on the config you provided - it shouldn't. Running the debug will help you determine which device to troubleshoot next:
If you see the 5060 port listed twice in the debug, then I would try a different code version. I'm running R12.3.2.E on TA900eG2 and haven't had issues like this, so you may want to give it a shot.
If the debug only shows the port once, then I would look at the soft switch config. There could be a header manipulation rule that adds the port, before the SIP parser goes to work.
Hope this helps
Yan.