News & Reviews
More How-To's & Tips More News
More Reviews Device Configuration Tools
No account yet? Create one
Forgot your Username or Password?

Welcome to the Voxilla VoIP Forum.

Voxilla has been a trusted source for accurate, up-to-date information on the IP Communications industry since 2002. A dedicated staff of reporters and engineers produce feature articles and product reviews to keep industry watchers abreast of the people, companies, and trends driving a fast moving market.

You are currently viewing our boards as a guest which gives you limited access to view most discussions and access our other features. By joining our free community you will have access to post topics, communicate privately with other members (PM), respond to polls, upload content and access many other special features. Registration is fast, simple and absolutely free so please, join our community today!

If you have any problems with the registration process or your account login, please contact contact us.





Closed Thread
 
LinkBack Thread Tools Rate Thread Display Modes
  #1 (permalink)  
Old January 26th, 2007, 06:36 AM
marianoacc marianoacc is offline
Junior Member
 
Join Date: Jan 2007
Posts: 6
marianoacc is on a distinguished road
Default SPA3102 FXS Line 1 dont work with Asterisk

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.
Digg this Post!Add Post to del.icio.usBookmark Post in TechnoratiFurl this Post!
Closed Thread


Thread Tools
Display Modes Rate This Thread
Rate This Thread:



Similar Threads for: SPA3102 FXS Line 1 dont work with Asterisk
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
SPA3102 with multiple phones connected to FXS cavallad Linksys (Sipura) VoIP Support Forum 3 January 11th, 2007 03:12 PM
SPA-3102 - FXS Port Impedance Carrot Cruncher Linksys (Sipura) VoIP Support Forum 1 November 15th, 2006 08:01 PM
It work on Sipura SPA3K but Problems on Linksys SPA3102 nish Linksys (Sipura) VoIP Support Forum 0 July 4th, 2006 03:38 AM
Sipura SPA-3102 FXS to FXO Problem zone Linksys (Sipura) VoIP Support Forum 7 June 19th, 2006 05:42 PM



All times are GMT. The time now is 09:08 PM.


vBulletin, Copyright ©2000 - 2008, Jelsoft Enterprises Ltd. SEO by vBSEO 3.0.0 ©2007, Crawlability, Inc. Logos and trademarks are the property of Voxilla or their respective owner. All other content © 2003-2007 by Voxilla, Inc.