Re: [asterisk-users] Sipura SPA3000
Larry, There is later firmware 3.1.10 dated March 2006. I gave up on the SPA3K. I could not solve the echo problems. Rich Adamson indicated that the SPA3K did not have logic to fall back to the PSTN on SIP failure, only loss of link to the network. I would have thought that Sipura could have used a registration failure as a SIP failure. The registration time would need to be relatively short to detect a failure in a reasonable amount of time. To test registration failover to PSTN, change the password for the SPA3K on the asterisk side. That should cause a registration failure. Then try placing a call to the PSTN line from elsewhere (a cell). Try calling out as well. Neither Linksys nor Sipura seem to have updated manuals. Try the latest firmware, but get a copy of the current, just in case. Bob... On Wed, 2006-11-15 at 23:22 -0600, Larry Alkoff wrote: Bob I have a further question about Fallback: On my Line 1 tab, the last item is VoIP Fallback To PSTN but there is no setting that can be changed. I _think_ my firmware is the latest when I bought the unit Oct 25, 2005 but _possibly_ I have no fallback. Bummer. Could you comment on this please? BTW, section 4.11 talks only about pstn calls ringing line 1: The voice path is (7) (6) (4) (2) (1). This feature is enabled by setting PSTN Ring Thru Line 1 to “yes”. If enabled, all incoming PSTN calls will ring the Line 1 phone regardless the VoIP gateway function is enabled on the SPA or not. Hence the same phone can be used to receive calls from Line 1 VoIP and from the PSTN. Section 4.9 talks about Fallback to PSTN but I'm not sure how to test this with my setup or even if fallback is implemented in my SPA3k: 4.9. Line 1 VoIP Fallback to PSTN When power is removed from the SPA-3000, the FXS port will be connected to the FXO port. In this case, the telephone attached to the FXS port is electrically connected to the PSTN service via the FXO port. When power is applied to the SPA, the FXS port will be disconnected from the FXO port. However, if the PSTN line is in use when the power is applied to the SPA, the relay will not be flipped until the PSTN line is released. This is done so that the SPA will not interrupt any call in progress on the PSTN line. When Line 1 VoIP service is down (due to registration failure or loss of Ethernet link), SPA can be configured to automatically route all outbound calls to the internal gateway if Auto PSTN Fallback ([Line 1] tab) is set to “yes”. The PSTN gateway applies the Line 1 Fallback DP to further limit the calls that can be made by the Line 1 caller during the fallback operation; this dial plan may be set to “none”. This case also belongs to call type #7 and the voice path is (1) (2) (4) (6) (7). Of course, I'm having a lot of trouble reading this complex manual g Larry Bob Chiodini wrote: Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22 By default, if my asterisk went down after the SPA3000 was already registered, the in-bound PSTN call was lost. I probably did not wait long enough and I did not have PSTN Call Ring Thru Line 1 enabled. Bob... On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote: My 3000 does this natively without config. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura SPA3000
Bob I have a further question about Fallback: On my Line 1 tab, the last item is VoIP Fallback To PSTN but there is no setting that can be changed. I _think_ my firmware is the latest when I bought the unit Oct 25, 2005 but _possibly_ I have no fallback. Bummer. Could you comment on this please? BTW, section 4.11 talks only about pstn calls ringing line 1: The voice path is (7) (6) (4) (2) (1). This feature is enabled by setting PSTN Ring Thru Line 1 to “yes”. If enabled, all incoming PSTN calls will ring the Line 1 phone regardless the VoIP gateway function is enabled on the SPA or not. Hence the same phone can be used to receive calls from Line 1 VoIP and from the PSTN. Section 4.9 talks about Fallback to PSTN but I'm not sure how to test this with my setup or even if fallback is implemented in my SPA3k: 4.9. Line 1 VoIP Fallback to PSTN When power is removed from the SPA-3000, the FXS port will be connected to the FXO port. In this case, the telephone attached to the FXS port is electrically connected to the PSTN service via the FXO port. When power is applied to the SPA, the FXS port will be disconnected from the FXO port. However, if the PSTN line is in use when the power is applied to the SPA, the relay will not be flipped until the PSTN line is released. This is done so that the SPA will not interrupt any call in progress on the PSTN line. When Line 1 VoIP service is down (due to registration failure or loss of Ethernet link), SPA can be configured to automatically route all outbound calls to the internal gateway if Auto PSTN Fallback ([Line 1] tab) is set to “yes”. The PSTN gateway applies the Line 1 Fallback DP to further limit the calls that can be made by the Line 1 caller during the fallback operation; this dial plan may be set to “none”. This case also belongs to call type #7 and the voice path is (1) (2) (4) (6) (7). Of course, I'm having a lot of trouble reading this complex manual g Larry Bob Chiodini wrote: Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22 By default, if my asterisk went down after the SPA3000 was already registered, the in-bound PSTN call was lost. I probably did not wait long enough and I did not have PSTN Call Ring Thru Line 1 enabled. Bob... On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote: My 3000 does this natively without config. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura SPA3000
Hi all,Im quite new to SPA3000. I have a TRIXBOX running on public address. I need my SPA3000's FXO to be used as a trunk from a dynamic address behind NAT. Is this scenario possible?Please give me some good links if it works.. I would really appreciate any help as my TRIXBOX is in US and my SPA3000 in middle east. Thanks everyoneDan.On 02/09/06, Rich Adamson [EMAIL PROTECTED] wrote: That option addresses what to do with the fxs (line 1) when theregistration fails as opposed to what does the fxo (pstn line) does when registration fails.Bob Chiodini wrote: Rich, After reading a little more, how about the Line 1 VoIP Fallback to PSTN (section 4.9)?It looks like this is invoked when the Ethernet link is down or registration fails.I don't have a SPA3000 up at the moment to look at what's required. Bob... On Fri, 2006-09-01 at 11:45 -0500, Rich Adamson wrote: If pstn call ring thru line 1 is enabled, all incoming pstn calls will ring through to the fxs port (and not to asterisk). The OP was looking for a auto fail over function that essentially would be pstn call ring thru line 1 on sip failure. That doesn't exist. Bob Chiodini wrote: Probably the PSTN Call Ring Thru Line 1 feature.Section 4.11 in: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22 By default, if my asterisk went down after the SPA3000 was already registered, the in-bound PSTN call was lost.I probably did not wait long enough and I did not have PSTN Call Ring Thru Line 1 enabled. Bob... On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote: My 3000 does this natively without config. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (includingSipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura SPA3000
I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura SPA3000
On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sipura SPA3000
My 3000 does this natively without config. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura SPA3000
Steve Kennedy wrote: On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. On power failure, yes. On ethernet cable disconnect, yes. But, when asterisk simply does not respond (for any reason), no. The last part is the difficult part. There isn't any logic in the spa3k that would essentially ping the asterisk service to see if it responds, and then do some alternate action if it does not respond. (One easy way to confirm that is to look around the spa3k config and see if you can find anything that relates to sip failure fail-over, timing entries associated with detecting a sip failure (lack of response), etc.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sipura SPA3000
Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22 By default, if my asterisk went down after the SPA3000 was already registered, the in-bound PSTN call was lost. I probably did not wait long enough and I did not have PSTN Call Ring Thru Line 1 enabled. Bob... On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote: My 3000 does this natively without config. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura SPA3000
If pstn call ring thru line 1 is enabled, all incoming pstn calls will ring through to the fxs port (and not to asterisk). The OP was looking for a auto fail over function that essentially would be pstn call ring thru line 1 on sip failure. That doesn't exist. Bob Chiodini wrote: Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22 By default, if my asterisk went down after the SPA3000 was already registered, the in-bound PSTN call was lost. I probably did not wait long enough and I did not have PSTN Call Ring Thru Line 1 enabled. Bob... On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote: My 3000 does this natively without config. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura SPA3000
That option addresses what to do with the fxs (line 1) when the registration fails as opposed to what does the fxo (pstn line) does when registration fails. Bob Chiodini wrote: Rich, After reading a little more, how about the Line 1 VoIP Fallback to PSTN (section 4.9)? It looks like this is invoked when the Ethernet link is down or registration fails. I don't have a SPA3000 up at the moment to look at what's required. Bob... On Fri, 2006-09-01 at 11:45 -0500, Rich Adamson wrote: If pstn call ring thru line 1 is enabled, all incoming pstn calls will ring through to the fxs port (and not to asterisk). The OP was looking for a auto fail over function that essentially would be pstn call ring thru line 1 on sip failure. That doesn't exist. Bob Chiodini wrote: Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22 By default, if my asterisk went down after the SPA3000 was already registered, the in-bound PSTN call was lost. I probably did not wait long enough and I did not have PSTN Call Ring Thru Line 1 enabled. Bob... On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote: My 3000 does this natively without config. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sipura SPA3000
Hello everyone, I have a question on configuration of SPA3000 with asterisk.I want all incoming calls are redirected from SPA3000 to my asterisk server.Asterisk then should direct this call to my SIP phones (including Sipura) In case asterisk server is down I want that call be directed straight to the handset connected to the SipuraIs this configuration possible?Thanks in advance,Michael Strelnikov ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura SPA3000
Michael Strelnikov wrote: 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? Yes for 1 and 2, never tested the #3. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
