badgewick Posted November 9, 2007 Report Share Posted November 9, 2007 I've finally gotten the CS410 connected the way we wanted it and it is working. However we've noticed a delay in the calls coming through from when we had the analog phones connected. Is this normal behavior or is there something I can change? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted November 9, 2007 Report Share Posted November 9, 2007 Could it be one ring? Maybe the PBX is waiting for the caller-ID, which is sent between the first and the second ring. Quote Link to comment Share on other sites More sharing options...
schneider Posted December 7, 2007 Report Share Posted December 7, 2007 I'm having the same problem... I also don't get CID (Australian install) it can take five or so seconds before the first ring... Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 7, 2007 Report Share Posted December 7, 2007 Well the caller-ID is transmitted between the first and the second ring. That's why analog phones with Caller-ID support first start ringing and then after that start showing the Caller-ID. Unfortunately, the genius that specified SIP did not envision that and wrote that the caller-ID cannot be changed after call setup (though there is a new RFC that now starts supporting that, but procatically all SIP phones not support that yet). That is why the CS410 FXO gateway waits for the second ring before it sends the call to the SIP phone. Quote Link to comment Share on other sites More sharing options...
edwardforgacs Posted July 29, 2009 Report Share Posted July 29, 2009 Well the caller-ID is transmitted between the first and the second ring. That's why analog phones with Caller-ID support first start ringing and then after that start showing the Caller-ID. Unfortunately, the genius that specified SIP did not envision that and wrote that the caller-ID cannot be changed after call setup (though there is a new RFC that now starts supporting that, but procatically all SIP phones not support that yet). That is why the CS410 FXO gateway waits for the second ring before it sends the call to the SIP phone. Sorry to re-open an old thread, but we are using this device in an Australian setup and have the same problem. As we don't even have caller ID enabled on the PSTN line as you have to pay for it (we use PSTN only to retain an old number and for emergency calls), can we disable the caller ID functionality in pbxnsip for that line to avoid the delay? Quote Link to comment Share on other sites More sharing options...
pbx support Posted July 29, 2009 Report Share Posted July 29, 2009 Sorry to re-open an old thread, but we are using this device in an Australian setup and have the same problem. As we don't even have caller ID enabled on the PSTN line as you have to pay for it (we use PSTN only to retain an old number and for emergency calls), can we disable the caller ID functionality in pbxnsip for that line to avoid the delay? You can not disable functionality today. But we may add a setting on the PSTN config page in 4.0 version. Quote Link to comment Share on other sites More sharing options...
pbx support Posted July 29, 2009 Report Share Posted July 29, 2009 You can not disable functionality today. But we may add a setting on the PSTN config page in 4.0 version. We have updated the binary to support this. Could you please PM to support@pbxnsip.com for the details? Also, please send the ssh information to the system(if possible) in the PM. Quote Link to comment Share on other sites More sharing options...
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