Re: [asterisk-users] asterisk + cisco 3825 with ISDN
oh so sorry about this. i just thought maybe someone had experienced the same. sorry again. regards Ron On 8/24/10 10:28 PM, David Backeberg wrote: > On Tue, Aug 24, 2010 at 9:05 AM, Ron wrote: >> hi all, >> >> i recently subscribe for an isdn and terminate it on a 3825 router. >> >> i used it as a sip trunk for my asterisk. i'm a newbie when it comes to >> ISDN. and i've been experiencing some issues: >> >> 1. Call Hangup: >> >> When hangup is initiated from the outside the extension (softphone/ip >> phone) does not hangup, is this normal? shouldn't asterisk hangup the >> extension as well when it receives the hangup properly from ISDN? or >> maybe it's because asterisk does not detect the proper hangup? > > I don't understand your inside versus outside description. But the main point: > * you have ISDN on a 3825, and you have voip. > * somewhere the SIP hangup doesn't initiate an ISDN hangup. > > This may be (is probably) due to you having your ISDN settings > incorrect for the line setup. There are about a dozen combinations of > settings for the way your ISDN line could be setup. > > Is this a T1 PRI? > Are you taking timing from the telco? > Do you have your D-channel set correctly? > Do you have your DSP settings correct for your chosen voip codec? > > You need an IOS manual to make sure all that is set correctly. You > can't download one from Cisco unless you pay for support and get a > service contract. Once you get that right, you'll also see the > settings in the Cisco manual for pushing caller ID correctly. I think > all of your problems are on the Cisco side, and zero are on the > asterisk side. > > Or you could scrap the 3825 and get a Digium PRI card and use DAHDI. > Then it would be an asterisk question :) > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk + cisco 3825 with ISDN
On Tue, Aug 24, 2010 at 9:05 AM, Ron wrote: > hi all, > > i recently subscribe for an isdn and terminate it on a 3825 router. > > i used it as a sip trunk for my asterisk. i'm a newbie when it comes to > ISDN. and i've been experiencing some issues: > > 1. Call Hangup: > > When hangup is initiated from the outside the extension (softphone/ip > phone) does not hangup, is this normal? shouldn't asterisk hangup the > extension as well when it receives the hangup properly from ISDN? or > maybe it's because asterisk does not detect the proper hangup? I don't understand your inside versus outside description. But the main point: * you have ISDN on a 3825, and you have voip. * somewhere the SIP hangup doesn't initiate an ISDN hangup. This may be (is probably) due to you having your ISDN settings incorrect for the line setup. There are about a dozen combinations of settings for the way your ISDN line could be setup. Is this a T1 PRI? Are you taking timing from the telco? Do you have your D-channel set correctly? Do you have your DSP settings correct for your chosen voip codec? You need an IOS manual to make sure all that is set correctly. You can't download one from Cisco unless you pay for support and get a service contract. Once you get that right, you'll also see the settings in the Cisco manual for pushing caller ID correctly. I think all of your problems are on the Cisco side, and zero are on the asterisk side. Or you could scrap the 3825 and get a Digium PRI card and use DAHDI. Then it would be an asterisk question :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk + cisco 3825 with ISDN
hi all, i recently subscribe for an isdn and terminate it on a 3825 router. i used it as a sip trunk for my asterisk. i'm a newbie when it comes to ISDN. and i've been experiencing some issues: 1. Call Hangup: When hangup is initiated from the outside the extension (softphone/ip phone) does not hangup, is this normal? shouldn't asterisk hangup the extension as well when it receives the hangup properly from ISDN? or maybe it's because asterisk does not detect the proper hangup? 2. Caller-ID for incoming calls: When i call in to the ISDN using my hand phone there are times that my handphone number shows up on the extension but there are times it does not, what it shows is the main isdn number i subscribed to. please sniff below: Correct Caller-ID: U 202.202.202.203:59049 -> 202.202.202.202:5060 INVITE sip:82301...@202.202.202.202:5060 SIP/2.0..Via: SIP/2.0/UDP 202.202.202.203:5060;branch=z9hG4bK53D21C6..Remote-Party-ID: ;party=calling;scre en=yes;privacy=off..From: ;tag=AD146FC4-F34..To: ..# i called using my mobile (33232333) to 82301000 on the extension i see the correct caller-id 33232333. Wrong Caller-ID: U 202.202.202.203:59123 -> 202.202.202.202:5060 INVITE sip:82301...@202.202.202.202:5060 SIP/2.0..Via: SIP/2.0/UDP 202.202.202.203:5060;branch=z9hG4bK53D51EE1..Remote-Party-ID: ;party=calling;scr een=yes;privacy=off..From: ;tag=AD14FB28-1D36..To: .. i called using my mobile (33232333) to 82301000 on the extension i see the correct caller-id 82301510 which is wrong (this number is the main line for my isdn) Can anyone please advice ow do i start troubleshooting this issues? TIA. Regards Ron -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
