AG1 Posted May 15, 2013 Report Share Posted May 15, 2013 I am reading through the Snom One book and I see I should be able to make entries in a dial plan in tandem to ensure a call goes through even if I lose my internet connection. I currently have 2 SIP trunks and I also have a Sangoma gateway with 4 FXO ports. In the book they use 1978* in the Pattern field and again in the Replacement field Is this the correct code or pattern to use to accomplish redundancy? Quote Link to comment Share on other sites More sharing options...
AG1 Posted May 15, 2013 Author Report Share Posted May 15, 2013 Also I have noticed that since I had some problems a week ago while we upgraded firmware on my Sangoma A200 I cannot call out on my VoIP lines/trunks. Incoming works fine ????? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 15, 2013 Report Share Posted May 15, 2013 I am reading through the Snom One book and I see I should be able to make entries in a dial plan in tandem to ensure a call goes through even if I lose my internet connection. I currently have 2 SIP trunks and I also have a Sangoma gateway with 4 FXO ports. In the book they use 1978* in the Pattern field and again in the Replacement field Is this the correct code or pattern to use to accomplish redundancy? It is important that the preference numbers in the dial plan are distinct, because the processing continues with the next higher preference after the one that triggered the failover. If you have set the country code to "1" then you should be using the pattern "978*". Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 15, 2013 Report Share Posted May 15, 2013 It is important that the preference numbers in the dial plan are distinct, because the processing continues with the next higher preference after the one that triggered the failover. If you have set the country code to "1" then you should be using the pattern "978*". That most probably depends on the SIP Header setup. Try to use "No Indication" in the trunk for the Sangoma card. This is absolutely okay as the analog line has the caller-ID tied to the cable anyway. Quote Link to comment Share on other sites More sharing options...
comspec Posted July 23, 2013 Report Share Posted July 23, 2013 I have a similar problem. I'm trying to setup a Vega 50 so that in the event of a SIP failure the customer can make inbound and outbound. Currently the Vega 50 is sending the POTS fine to the PBX, so inbound is fine when the trunk goes down. The trouble I'm having is trying to work-out how to get the PBX to dial out when the SIP Out fails. Currently I'm getting "Request Timeout". I have the following settings which may shed some light on what is wrong if you can help? Trunk A Routing Destination for incoming calls: Sends calls to... Failover behaviour: Always, except.... Request Timeout: 10 Accept Redirect: No Error message: 500 Line.... Trunk B (Vegastream) Routing Destination for incomming calls: Send all calsl to... Default account: xxx Failover Timeout: 10 Accept Redirect: No Error message..:500 line... Redirect destination.. : 0201 (this is the FXO port) Dialplan Name: Outbound Call Dialplan Pref 100 Trunk A Pattern * Service Flag unassigned Pref 300 Trunk B Pattern * Service Flag unassigned Standard Dialplan Pref 200 Trunk A Pattern * Service Flag 9am-5pm M-F Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted July 23, 2013 Report Share Posted July 23, 2013 At first glance this looks good. I would not use any timeouts on the Vegastream (the Ethernet connection to the Vega is not supposed to go down). But that is not the problem I guess. Next step is to dig deeper into the logs. Make sure that you have "trunks" set for logging on level 8 or 9... Quote Link to comment Share on other sites More sharing options...
comspec Posted July 24, 2013 Report Share Posted July 24, 2013 Here are the logs when trying to make an outbound call. As you say it's almost there but until I can make an outbound call this option of failover with the Vega 50 isn't an option just yet. Your help as always is greatly appreciated. [5] 2013/07/24 12:10:15: Identify trunk (domain name match) 4 [8] 2013/07/24 12:10:15: To is <sip:##########@pbx.mydomain.com>, user 0, domain 1 [9] 2013/07/24 12:10:15: Generating hf header using {from} [9] 2013/07/24 12:10:15: Generating ht header using {to} [9] 2013/07/24 12:10:15: Generating hpai header using {trunk} [9] 2013/07/24 12:10:15: Generating hf header using {trunk} [9] 2013/07/24 12:10:15: Generating ht header using {request-uri} Quote Link to comment Share on other sites More sharing options...
AG1 Posted August 19, 2013 Author Report Share Posted August 19, 2013 I still cannot get my 870's to dial out on my Nextiva SIP trunks The 820s work fine so it has to be an 870 firmware issue....does it not? If so does the beta version 8.7.4.8 fix this problem? Is anyone else having this problem with 870's or am I just the lucky one again? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted August 19, 2013 Report Share Posted August 19, 2013 Eh is that related to the topic? Opening new topics does not cost anything... Quote Link to comment Share on other sites More sharing options...
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