hank smith
2005-May-23 13:32 UTC
[Asterisk-Users] spa-1001 not getting a dial tone on my pbx
hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology Inc Info System SIP Provisioning Regional Line 1 User 1 User Login basic | advanced Table with 4 columns and 75 rows Line Enable: yes Streaming Audio Server (SAS) SAS Enable: no SAS DLG Refresh Intvl: 30 SAS Inbound RTP Sink: NAT Settings NAT Mapping Enable: yes NAT Keep Alive Enable: yes NAT Keep Alive Msg: $NOTIFY NAT Keep Alive Dest: $PROXY Network Settings SIP TOS/DiffServ Value: 0x68 Network Jitter Level: high RTP TOS/DiffServ Value: 0xb8 SIP Settings SIP Port: 5060 SIP 100REL Enable: no EXT SIP Port: Auth Resync-Reboot: yes SIP Debug Option: none Call Feature Settings Blind Attn-Xfer Enable: no MOH Server: Xfer When Hangup Conf: yes Proxy and Registration Proxy: 67.183.118.6 Use Outbound Proxy: no Outbound Proxy: Use OB Proxy In Dialog: no Register: yes Make Call Without Reg: no Register Expires: 60 Ans Call Without Reg: no Use DNS SRV: no DNS SRV Auto Prefix: no Proxy Fallback Intvl: 3600 Subscriber Information Display Name: Herbie Allen User ID: 202 Password: ************* Use Auth ID: yes Auth ID: 202 Mini Certificate: SRTP Private Key: Supplementary Service Subscription Call Waiting Serv: yes Block CID Serv: yes Block ANC Serv: yes Dist Ring Serv: yes Cfwd All Serv: yes Cfwd Busy Serv: yes Cfwd No Ans Serv: yes Cfwd Sel Serv: yes Cfwd Last Serv: yes Block Last Serv: yes Accept Last Serv: yes DND Serv: yes CID Serv: yes CWCID Serv: yes Call Return Serv: yes Call Back Serv: yes Three Way Call Serv: yes Three Way Conf Serv: yes Attn Transfer Serv: yes Unattn Transfer Serv: yes MWI Serv: yes VMWI Serv: yes Speed Dial Serv: yes Secure Call Serv: yes Referral Serv: yes Feature Dial Serv: yes Audio Configuration Preferred Codec: G711u Silence Supp Enable: no Use Pref Codec Only: no Silence Threshold: medium G729a Enable: yes Echo Canc Enable: yes G723 Enable: yes Echo Canc Adapt Enable: yes G726-16 Enable: yes Echo Supp Enable: yes G726-24 Enable: yes FAX CED Detect Enable: yes G726-32 Enable: yes FAX CNG Detect Enable: yes G726-40 Enable: yes FAX Passthru Codec: G711u DTMF Tx Method: Auto FAX Codec Symmetric: yes Hook Flash Tx Method: None FAX Passthru Method: NSE Release Unused Codec: yes FAX Process NSE: yes Dial Plan Dial Plan: (xx.|*xx.|**xx.|#xx.) Enable IP Dialing: no FXS Port Polarity Configuration Idle Polarity: Forward Caller Conn Polarity: Forward Callee Conn Polarity: Forward table end Undo All Changes Submit All Changes User Login basic | advanced Copyright ? 2003 Sipura Technology. All Rights Reserved. -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 5/22/2005 email: hanksmith4@earthlink.net gmail: hanksmith5@gmail.com msn messenger: hanksmith4@earthlink.net aim: hanksmith5 skype: hanksmith5 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/dd7173ec/attachment.htm
Seemingly Similar Threads
- spa-1001 with asterisk?
- having asterisk play music on hold to callers while phone rings?
- How to configure a Linksys PAP2T ATA to connect an analog fax machine to Asterisk
- adding up elements within a list
- Re: [Asterisk-Dev] ANNOUNCE: CLASS-like features for Asterisk