Below my settings (SIP & Line1 tab of the PAP2t configuration page, les.net trunk setup). Which way is your one-way audio? I recently observed one-way audio (no outgoing audio!) with another provider which is registered in my voxalot account.
PAP2T-SIP:
Code:
SIP
Parameters
Max [70 ] Max [5 ]
Forward: Redirection:
Max Auth: [2 ] SIP User [$VERSION ]
Agent Name:
SIP Server [$VERSION ] SIP Reg User [ ]
Name: Agent Name:
SIP Accept [ ] DTMF Relay [application/dtmf-relay]
Language: MIME Type:
Hook Flash [application/hook-flash] Remove Last [no ]
MIME Type: Reg:
Use Escape
Compact [no ] Display [no ]
Header: Name:
SIP Timer
Values
(sec)
SIP T1: [.5 ] SIP T2: [4 ]
SIP T4: [5 ] SIP Timer B: [32 ]
SIP Timer [32 ] SIP Timer H: [32 ]
F:
SIP Timer [32 ] SIP Timer J: [32 ]
D:
INVITE [240 ] ReINVITE [30 ]
Expires: Expires:
Reg Min [1 ] Reg Max [7200 ]
Expires: Expires:
Reg Retry [30 ] Reg Retry [1200 ]
Intvl: Long Intvl:
Response
Status
Code
Handling
SIT1 RSC: [ ] SIT2 RSC: [ ]
SIT3 RSC: [ ] SIT4 RSC: [ ]
Try Backup [ ] Retry Reg [ ]
RSC: RSC:
RTP
Parameters
RTP Port [16384 ] RTP Port [16482 ]
Min: Max:
RTP Packet [0.030 ] Max RTP ICMP [0 ]
Size: Err:
RTCP Tx [0 ] No UDP [no ]
Interval: Checksum:
Stats In [no ]
BYE:
SDP
Payload
Types
NSE AVT Dynamic
Dynamic [100 ] Payload: [101 ]
Payload:
INFOREQ G726r16
Dynamic [ ] Dynamic [98 ]
Payload: Payload:
G726r24 G726r40
Dynamic [97 ] Dynamic [96 ]
Payload: Payload:
G729b NSE Codec
Dynamic [99 ] Name: [NSE ]
Payload:
AVT Codec [telephone-event] G711u Codec [PCMU ]
Name: Name:
G711a G726r16
Codec [PCMA ] Codec Name: [G726-16 ]
Name:
G726r24 G726r32
Codec [G726-24 ] Codec Name: [G726-32 ]
Name:
G726r40 G729a Codec
Codec [G726-40 ] Name: [G729a ]
Name:
G729b G723 Codec
Codec [G729ab ] Name: [G723 ]
Name:
NAT
Support
Parameters
Handle VIA [no ] Handle VIA [no ]
received: rport:
Insert VIA [no ] Insert VIA [no ]
received: rport:
Substitute [yes] Send Resp To [no ]
VIA Addr: Src Port:
STUN [yes] STUN Test [no ]
Enable: Enable:
STUN [stun.voxalot.com.au] EXT IP: [ ]
Server:
EXT RTP [ ] NAT Keep [15 ]
Port Min: Alive Intvl:
PAP2T-Line1:
Code:
Streaming
Audio Server
(SAS)
SAS Enable: [no ] SAS DLG Refresh [30 ]
Intvl:
SAS Inbound [ ]
RTP Sink:
NAT Settings
NAT Mapping [yes] NAT Keep Alive [yes]
Enable: Enable:
NAT Keep Alive [$NOTIFY ] NAT Keep Alive [$PROXY ]
Msg: Dest:
Network
Settings
SIP TOS/ Network Jitter
DiffServ [0x68 ] Level: [high ]
Value:
RTP TOS/ Jitter Buffer
DiffServ [0xb8 ] Adjustment: [up and down]
Value:
SIP Settings
SIP Port: [5060 ] SIP 100REL [no ]
Enable:
EXT SIP Port: [ ] Auth [yes]
Resync-Reboot:
SIP [ ] SIP [no ]
Proxy-Require: Remote-Party-ID:
SIP GUID: [no ] SIP Debug [1-line ]
Option:
RTP Log Intvl: [0 ] Restrict Source [no ]
IP:
Referor Bye [4 ] Refer Target Bye [0 ]
Delay: Delay:
Referee Bye [0 ] Refer-To Target [no ]
Delay: Contact:
Sticky 183: [no ]
Call Feature
Settings
Blind
Attn-Xfer [no ] MOH Server: [ ]
Enable:
Xfer When [yes] Conference [ ]
Hangup Conf: Bridge URL:
Conference [3 ]
Bridge Ports:
Proxy and
Registration
Proxy: [us.voxalot.com ] Use Outbound [no ]
Proxy:
Outbound [ ] Use OB Proxy In [no ]
Proxy: Dialog:
Register: [yes] Make Call [no ]
Without Reg:
Register [600 ] Ans Call Without [yes]
Expires: Reg:
Use DNS SRV: [yes] DNS SRV Auto [yes]
Prefix:
Proxy Fallback [600 ] Proxy Redundancy [Based on SRV Port]
Intvl: Method:
Voice Mail [us.voxalot.com ]
Server:
Subscriber
Information
Display Name: [ABCD ] User ID: [nnnnnn ]
Password: [************* ] Use Auth ID: [yes]
Auth ID: [nnnnnn ]
Mini [ ]
Certificate:
SRTP Private [ ]
Key:
Supplementary
Service
Subscription
Call Waiting [yes] Block CID Serv: [yes]
Serv:
Block ANC [yes] Dist Ring Serv: [yes]
Serv:
Cfwd All Serv: [yes] Cfwd Busy Serv: [yes]
Cfwd No Ans [yes] Cfwd Sel Serv: [yes]
Serv:
Cfwd Last [yes] Block Last Serv: [yes]
Serv:
Accept Last [yes] DND Serv: [yes]
Serv:
CID Serv: [yes] CWCID Serv: [yes]
Call Return [yes] Call Back Serv: [yes]
Serv:
Three Way Call [yes] Three Way Conf [yes]
Serv: Serv:
Attn Transfer [yes] Unattn Transfer [yes]
Serv: Serv:
MWI Serv: [yes] VMWI Serv: [yes]
Speed Dial [yes] Secure Call [yes]
Serv: Serv:
Referral Serv: [yes] Feature Dial [yes]
Serv:
Service
Announcement [no ]
Serv:
Audio
Configuration
Preferred [G729a ] Silence Supp [no ]
Codec: Enable:
Use Pref Codec [no ] Silence [medium]
Only: Threshold:
G729a Enable: [yes] Echo Canc [yes]
Enable:
G723 Enable: [yes] Echo Canc Adapt [yes]
Enable:
G726-16 [yes] Echo Supp [yes]
Enable: Enable:
G726-24 [yes] FAX CED Detect [yes]
Enable: Enable:
G726-32 [yes] FAX CNG Detect [yes]
Enable: Enable:
G726-40 [yes] FAX Passthru [G711u]
Enable: Codec:
FAX Codec [yes] FAX Passthru [NSE ]
Symmetric: Method:
DTMF Tx [Auto ] FAX Process NSE: [yes]
Method:
Hook Flash Tx [None] FAX Disable [no ]
Method: ECAN:
Release Unused [yes]
Codec:
Dial Plan
Dial Plan: [(x.|x*x.|**x.|*x.|<#:*>xxx.) ]
Enable IP [no ] Emergency [ ]
Dialing: Number:
FXS Port
Polarity
Configuration
Idle Polarity: [Forward] Caller Conn [Forward]
Polarity:
Callee Conn [Forward]
Polarity:
Line Enable: [yes]
trunk setup @ les.net:
Code:
Peer Name NNNNNNNNNNNNNNN
Your Description [forward to voxalot]
[?]Peer Technology [SIP ]
[?]DTMF Mode [rfc2833]
[?]Re-Invite [no ]
[?]NAT [no ]
[?]Error Method [Verbal]
[?]Codecs[*]G.711 [*]G.729 [*]GSM [*]G.726
[?]Peer Type [URI ] (URI/PSTN Experimental)
[?]Peer Address [nnnnnn@voxalot.com ]
[?]Password [ ]
[?]Registered No
[?]Registered IP nnnnnn@voxalot.com
[?]Registration Expires No
[?]Outbound CallerID [ ]
[?]7-Digit Dialing [ ]
[?]7-Digit Area Code Prefix [ ]
[?]10-Digit Dialing, Prefix 1 [ ]