Re: [asterisk-users] Linksys SPA devices and CID
Quoting Tim Johnson <[EMAIL PROTECTED]>: > What do you have for your "PSTN Answer Delay" (in PSTN tab)? I had to > set mine between 3 to 5 to get reliable CID from the POTS line. This > was for a SPA3102, not a 3000. I've never had a 3000, but everyone > says they are nearly identical. I normally have 0 for both "PSTN Answer Delay" and "PSTN Ring Thru Delay". Increasing the latter has also been said to solve this problem. However, if I change both of these values to 5 it does add a noticeable delay before any phones ring, but the CID remains unavailable. Perhaps this is because where I live (in the Netherlands) the local telco always sends the CID first. Thanks anyway! Cheers, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA devices and CID
Quoting Jaap Winius <[EMAIL PROTECTED]>: > Quoting Tim Johnson <[EMAIL PROTECTED]>: > >> Your caller ID is probably being over-ridden by the settings in your >> sip.conf file. Remove the caller ID from your PSTN section of the >> sip.conf, and the CID should be passed on from the POTS line. > > That sounds like a good idea regardless. On the SPA3000 I've changed > the User ID to "PSTN", while the sip.conf now has the following entry: > > [4500] > ; SPA3000, PSTN line: incoming. > type=friend > host=dynamic > port=5061 > context=home-in > username=PSTN > secret=1234 > dtmfmode=rfc2833 > disallow=all > allow=ulaw > insecure=very > qualify=yes > > While still not a solution in my case, this is an improvement. CIDs > for incoming PSTN calls are now reported as "Unavailable", instead of > always being "4500". > > Thanks! > > Jaap What do you have for your "PSTN Answer Delay" (in PSTN tab)? I had to set mine between 3 to 5 to get reliable CID from the POTS line. This was for a SPA3102, not a 3000. I've never had a 3000, but everyone says they are nearly identical. Tim Johnson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA devices and CID
Quoting Tim Johnson <[EMAIL PROTECTED]>: > Your caller ID is probably being over-ridden by the settings in your > sip.conf file. Remove the caller ID from your PSTN section of the > sip.conf, and the CID should be passed on from the POTS line. That sounds like a good idea regardless. On the SPA3000 I've changed the User ID to "PSTN", while the sip.conf now has the following entry: [4500] ; SPA3000, PSTN line: incoming. type=friend host=dynamic port=5061 context=home-in username=PSTN secret=1234 dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very qualify=yes While still not a solution in my case, this is an improvement. CIDs for incoming PSTN calls are now reported as "Unavailable", instead of always being "4500". Thanks! Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA devices and CID
Quoting Jaap Winius <[EMAIL PROTECTED]>: > Hi list, > > After successfully configuring Linksys SPA3000 and SPA3102 devices as > Asterisk PSTN gateways, the only thing I can't get working is the PSTN > Caller ID. The analog and SIP phones I've used can both display CIDs > for internal calls, while the analog model also displays CIDs > correctly when attached directly to the PSTN line. However, when PSTN > calls come in via the SPA device, all I see is the SPA device CID > associated with the PSTN line; not the CID of the incoming call. > > The only SPA settings I know of that are supposed to enable the > passing on of PSTN CIDs are the "PSTN CID For VoIP CID" option (under > PSTN Line), which AFAIK must be set to "yes," and the "Caller ID > Method" (under Regional), which I must set to "ETSI DTMI With PR", or > else my analog phone will not display any CIDs when attached to the > SPA's FXS port. Yet, these settings have never led to any positive > results, despite attempts with different firmware versions on both > devices. > > Can anyone help? > > Thanks, > > Jaap Your caller ID is probably being over-ridden by the settings in your sip.conf file. Remove the caller ID from your PSTN section of the sip.conf, and the CID should be passed on from the POTS line. Tim Johnson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys SPA devices and CID
Hi list, After successfully configuring Linksys SPA3000 and SPA3102 devices as Asterisk PSTN gateways, the only thing I can't get working is the PSTN Caller ID. The analog and SIP phones I've used can both display CIDs for internal calls, while the analog model also displays CIDs correctly when attached directly to the PSTN line. However, when PSTN calls come in via the SPA device, all I see is the SPA device CID associated with the PSTN line; not the CID of the incoming call. The only SPA settings I know of that are supposed to enable the passing on of PSTN CIDs are the "PSTN CID For VoIP CID" option (under PSTN Line), which AFAIK must be set to "yes," and the "Caller ID Method" (under Regional), which I must set to "ETSI DTMI With PR", or else my analog phone will not display any CIDs when attached to the SPA's FXS port. Yet, these settings have never led to any positive results, despite attempts with different firmware versions on both devices. Can anyone help? Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
