Re: [Asterisk-Users] realtime sip confusion
snacktime wrote: In case someone else made the same mistake I did, and because I can't find this information posted anywhere, here is what I found out about realtime sip. You can use it to register UA's that are registering to asterisk, and you can use it for peer context's for outgoing calls, but you cannot use it for incoming calls from gateways you have registered with. I would have thought that when a call came in it would query either for the hostname of the gateway you registered with, or maybe the extension you registered as, but instead it looks up the username of the caller, which for incoming calls will usually be the caller id. It makes sense when you stop and think about it, but it's not exactly intuitive at first. Thanks for the note but why do you say it makes sense? If the username of the caller is used to identify a peer that seems really bad. If used this way then I'd have to define every number that is likely to call into my Asterisk box. Could you explain? Thanks Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime sip confusion
On 6/27/05, Steve Blair [EMAIL PROTECTED] wrote: snacktime wrote: In case someone else made the same mistake I did, and because I can't find this information posted anywhere, here is what I found out about realtime sip. You can use it to register UA's that are registering to asterisk, and you can use it for peer context's for outgoing calls, but you cannot use it for incoming calls from gateways you have registered with. I would have thought that when a call came in it would query either for the hostname of the gateway you registered with, or maybe the extension you registered as, but instead it looks up the username of the caller, which for incoming calls will usually be the caller id. It makes sense when you stop and think about it, but it's not exactly intuitive at first. Thanks for the note but why do you say it makes sense? If the username of the caller is used to identify a peer that seems really bad. If used this way then I'd have to define every number that is likely to call into my Asterisk box. Could you explain? It makes sense because it mirrors how sip.conf works, as opposed to doing something different. To me it looks like a limitation of SIP, whereas IAX was designed to work in a PBX environment. I'm not clear on a lot of this, but with the way SIP works I can't see an easy way to get the callerid and the called number without using some custom and/or little used headers. I would be very interested in hearing about how this is customarily done. Obviously the providers are getting that information from their upstream proxies, otherwise they wouldn't be able to route the calls. Why that information isn't passed downstream I don't know. Maybe it requires customization of * beyond what a provider wants to support? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime sip confusion
snacktime wrote: Trying to use realtime sip for the first time, and it's not working as expected. I have one user entry in the sip database. Everthing else is still in sip.conf. When I get an incoming call, this is the database query: SELECT * FROM ast_home_sip_realtime WHERE name = '+18003859169' The 800# is the caller id of the caller, which doesnt' make any sense to me. Is there any documentation about how realtime sip/iax actually work beyond just the schema's that are on the wiki? Chris Seems to me that your UA is sending that number as its SIP Username. You can look in /var/log/asterisk/debug for lots of RealTime info if using res_config_mysql. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime sip confusion
Chris Seems to me that your UA is sending that number as its SIP Username. You can look in /var/log/asterisk/debug for lots of RealTime info if using res_config_mysql. This was an incoming call via a DID. I can call from any phone and the query is always on the callerid. Part of my problem is I'm completely guessing on how sip realtime works, there is absolutely nothing I can find that say's 'this is what sip realtime does in a user/peer/friend context'. Also there is a bug where if a context has a dash, realtime splits the string on the dash and does two queries. I don't know why it's picking up the context's in the first place since I don't understand the logic. I do know that I have a couple of unique context names such as 'from-teliax' or 'voicepulse-out', and in mysql I see realtime making queries like the following: SELECT * from sip where name = 'from' SELECt * from sip where name = 'teliax' Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime sip confusion
In case someone else made the same mistake I did, and because I can't find this information posted anywhere, here is what I found out about realtime sip. You can use it to register UA's that are registering to asterisk, and you can use it for peer context's for outgoing calls, but you cannot use it for incoming calls from gateways you have registered with. I would have thought that when a call came in it would query either for the hostname of the gateway you registered with, or maybe the extension you registered as, but instead it looks up the username of the caller, which for incoming calls will usually be the caller id. It makes sense when you stop and think about it, but it's not exactly intuitive at first. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
