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SPA3102 FXS Line 1 dont work with AsteriskTechnical support, how-to guides, troubleshooting, and general assistance for Linksys hardware. |
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| Hi, I'have a problem with my Line1 configuration. Sipura reports Line1 as registered, but I can't make that my analog phone work with it. If I setup "Auto PSTN Fallback" every time, even as registered and with network connection to Asterisk (No NAT, no Firewall in my lan), do the Fallback when I hold up my analog phone. If I sniff network packets and restart Sipura, I can see register request for PSTN Userid but no tries for Line1 Userid. I never received dial tone from my analog phone if fallback is not configured, I can't make ip calls from analog phone either. I can make outbound calls from soft phones through PSTN line without problems . My line 1 is configured as: ************************************ Line Enable:yes NAT Settings NAT Mapping Enable:No NAT Keep Alive Enable:No (others all defaults) SIP Settings SIP Port:5060 SIP 100REL Enable: no EXT SIP Port: Auth Resync-Reboot: yes SIP Proxy-Require: SIP Remote-Party-ID: yes SIP GUID: no SIP Debug Option: none RTP Log Intvl:0 Restrict Source IP: no Referor Bye Delay:4 Refer Target Bye Delay: 0 Referee Bye Delay:0 Refer-To Target Contact: yes Sticky 183: no Call Feature Settings Blind Attn-Xfer Enable: no MOH Server: Xfer When Hangup Conf: yes Proxy and Registration Proxy: 10.4.14.190 Outbound Proxy: Use Outbound Proxy: no Use OB Proxy In Dialog: yes Register: yes Make Call Without Reg: no Register Expires: 3600 Ans Call Without Reg: no Use DNS SRV: no DNS SRV Auto Prefix: no Proxy Fallback Intvl: 3600 Proxy Redundancy Method: Normal Voice Mail Server: Mailbox Subscribe Expires: 2147483647 Subscriber Information Display Name: Telefono User ID: lhstel Password: ***** Use Auth ID: no Auth ID: 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 Redial 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 Service Announcement Serv: no Audio Configuration Preferred Codec: G729a 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 Process INFO: yes FAX Codec Symmetric: yes DTMF Process AVT: yes FAX Passthru Method: NSE DTMF Tx Method: Auto FAX Process NSE: yes Hook Flash Tx Method: None FAX Disable ECAN: no Release Unused Codec: yes FAX Enable T38: yes FAX T38 Redundancy: 1 FAX Tone Detect Mode: caller or callee Symmetric RTP: yes Gateway Accounts Gateway 1: GW1 NAT Mapping Enable: no GW1 Auth ID: GW1 Password: Gateway 2: GW2 NAT Mapping Enable: no GW2 Auth ID: GW2 Password: Gateway 3: GW3 NAT Mapping Enable: no GW3 Auth ID: GW3 Password: Gateway 4: GW4 NAT Mapping Enable: no GW4 Auth ID: GW4 Password: VoIP Fallback To PSTN Auto PSTN Fallback: yes Dial Plan Dial Plan: (S0<:s@10.4.14.190:5060>) Enable IP Dialing: no Emergency Number: FXS Port Polarity Configuration Idle Polarity: Forward Caller Conn Polarity: Reverse Callee Conn Polarity: Reverse **************************** In sip.conf I have: [lhs] type=peer host=dynamic context=incoming secret=***** port=5061 mailbox=lhs dtmfmode=rfc2833 canreinvite=yes allow=all insecure=very [lhstel] type=peer host=dynamic context=incoming secret=***** port=5060 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=g723.1 allow=g729 allow=gsm insecure=very [pstn-spa3k] type=peer auth=md5 host=10.4.14.189 port=5061 secret=******* username=asterisk fromuser=asterisk dtfmode=rfc2833 context=home insecure=very ************************ Can someone help me please Regards Mariano Last edited by marianoacc : January 26th, 2007 at 06:53 AM. |
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| Thread | Thread Starter | Forum | Replies | Last Post |
| PBX Connection to SPA-3102 does not work | szumlanski | Linksys (Sipura) VoIP Support Forum | 6 | February 19th, 2007 05:07 PM |
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| It work on Sipura SPA3K but Problems on Linksys SPA3102 | nish | Linksys (Sipura) VoIP Support Forum | 0 | July 4th, 2006 03:38 AM |
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