Re: [asterisk-users] DUNDi and reinvites...
I'm talking out my rear so someone please apply an attitude adjustment if I'm way off base. But, if you are using Dundi as a lookup engine it should know the contact information both endpoints and how to reach them perhaps not ONLY knowing how to comunicate via another asterisk box. Much like simply initializing a base dns infrastructure for the CPE devices. If the CPE devices are configured to accept SIP transactions from $domain or both asterisk servers server A should be able to send a invite directly to client B and bring up the inbound call. As far as the client knows it's still talking and placing outbound calls with server B. IE: Client A calls Client B Client A hits Serv A. Serv A does lookup finds it knows about Client B Serv A sends the call direct to Client B's IP. I'm assuming that both servers are acting as mirrors of eachother, in that voicemail and all that is a //shared// resource.. so if Client B rings unavail/busy that your serv A knows what to do with the call. In general as long as a client device knows to understand and accept sip messages from $host an inbound call does not have to come from the server they registered to. If you look at a linksys adapter this is one of the reasons they have that domain parameter which controls the list of hosts that are allowed to send SIP transactions to the unit. Am I wrong on this? The only other artifact I can think of is the fact of NAT traversal, where if client B that's to recieve the call is behind a NAT firewall and you are not doing port forwarding of the SIP signaling then ofcourse it won't get the call because server A has not established the NAT association. But assuming you are using a common 'sbc' or gatekeeper (ser) that box would know the association and things would be happy. On Jun 7, 2007, at 7:11 PM, Jared Smith wrote: On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote: That's all fine and good until it becomes the receiving phone, and the other phone (as an originator) also has canreinvite set to yes. Then, your back to both Asterisk servers being completely taken out of the loop again! While I haven't taken the time to actually try this, I might suggest that you could set up separate user and peer sections in sip.conf, so that you can handle inbound calls differently that outbound calls. -Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi and reinvites...
I don't know if this is possible, and I can't quite get my head around how to do it... If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: +---+ +---+ | A |-| B | /+---+ +---+\ / \ Phone1 Phone2 Is there a way configure re-invites in this situation so that either Asterisk A or B drops out of the call, and there's only one Asterisk box between Phone1 and Phone2? Like this... +---+ +---+ | A | | B | /+---+\+---+ / \ Phone1 ---Phone2 Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi and reinvites...
On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote: If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: Correct so far... although once the call is made, it's no longer a DUNDi question, and is simply a signalling question. (In other words, DUNDi is used for Phone 1 to figure out how to connect to Phone 2, but after it's figured that out, it's a normal SIP or IAX call between Asterisk A and Asterisk B.) Is there a way configure re-invites in this situation so that either Asterisk A or B drops out of the call, and there's only one Asterisk box between Phone1 and Phone2? Like this… Yes, as long as the protocols are all the same. If Phone1 is talking SIP to Asterisk A and Asterisk A is talking IAX to Asterisk B and Asterisk B is talking SIP to Phone 2, then it won't happen. But assuming everything is using the same transport, they'll happen. In fact, if re-invites are enabled on both Asterisk servers, and the two phones can communicate directly, you can re-invite *both* Asterisk servers out of the middle of the call. -Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DUNDi and reinvites...
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Thursday, June 07, 2007 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi and reinvites... On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote: If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: Correct so far... although once the call is made, it's no longer a DUNDi question, and is simply a signalling question. (In other words, DUNDi is used for Phone 1 to figure out how to connect to Phone 2, but after it's figured that out, it's a normal SIP or IAX call between Asterisk A and Asterisk B.) Hi Jared. Understood. Is there a way configure re-invites in this situation so that either Asterisk A or B drops out of the call, and there's only one Asterisk box between Phone1 and Phone2? Like this... Yes, as long as the protocols are all the same. If Phone1 is talking SIP to Asterisk A and Asterisk A is talking IAX to Asterisk B and Asterisk B is talking SIP to Phone 2, then it won't happen. But assuming everything is using the same transport, they'll happen. In fact, if re-invites are enabled on both Asterisk servers, and the two phones can communicate directly, you can re-invite *both* Asterisk servers out of the middle of the call. I figured the protocols would have to be the same. The phones are SIP based, so I tried to get DUNDi to work with SIP. That's where I hit snags. The INVITE coming from Asterisk 1 has the original phone's From: address, because it's much easier for Asterisk 2 to accept calls from Asterisk 1 based on the IP address. However, because the INVITE still has the original phones FROM: tag, Asterisk matches it against it's own copy of Phone 1's sip entry, rather than the entry for Asterisk 1, and then sends a 407 proxy Auth message back to the Asterisk 1, who doesn't know what to with it. Another, much uglier approach, is to change the From Address that Asterisk 1 sends the INVITE with. However, then we'd need to add extra SIP headers to the INVITE going out from Asterisk 1. Asterisk two would authenticate against those and pluck out the extra SIP headers to get the original caller id. I also tried setting the username/secret on Asterisk 1 for it's trunk to Asterisk 2, thinking that the From: and the auth credentials would be different, but Asterisk threw a fit when the From: did not match the digest id. What's wrong with that? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DUNDi and reinvites...
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Thursday, June 07, 2007 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi and reinvites... On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote: If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: Correct so far... although once the call is made, it's no longer a DUNDi question, and is simply a signalling question. (In other words, DUNDi is used for Phone 1 to figure out how to connect to Phone 2, but after it's figured that out, it's a normal SIP or IAX call between Asterisk A and Asterisk B.) Is there a way configure re-invites in this situation so that either Asterisk A or B drops out of the call, and there's only one Asterisk box between Phone1 and Phone2? Like this... Yes, as long as the protocols are all the same. If Phone1 is talking SIP to Asterisk A and Asterisk A is talking IAX to Asterisk B and Asterisk B is talking SIP to Phone 2, then it won't happen. But assuming everything is using the same transport, they'll happen. In fact, if re-invites are enabled on both Asterisk servers, and the two phones can communicate directly, you can re-invite *both* Asterisk servers out of the middle of the call. Jared, we also don't want to reinvite all the way down to the two phones communicating with each other. We want a single Asterisk system between them. I just reconfigured my setup to send calls from Asterisk 1 to Asterisk 2 with a callerid/From: different to the originating phone's, just to get to the point where I can set reinvites up. Let's just say we only configured the originating phone with canreinvite=yes, which hopefully means the originating phone would reinvite with the second Asterisk server. That's all fine and good until it becomes the receiving phone, and the other phone (as an originator) also has canreinvite set to yes. Then, your back to both Asterisk servers being completely taken out of the loop again! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi and reinvites...
Douglas Garstang wrote: Let's just say we only configured the originating phone with canreinvite=yes, which hopefully means the originating phone would reinvite with the second Asterisk server. That's all fine and good until it becomes the receiving phone, and the other phone (as an originator) also has canreinvite set to yes. Then, your back to both Asterisk servers being completely taken out of the loop again! reinvites only remove the server from the AUDIO PATH. Signaling is still going thru Asterisk no matter what happens. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DUNDi and reinvites...
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, June 07, 2007 2:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi and reinvites... Douglas Garstang wrote: Let's just say we only configured the originating phone with canreinvite=yes, which hopefully means the originating phone would reinvite with the second Asterisk server. That's all fine and good until it becomes the receiving phone, and the other phone (as an originator) also has canreinvite set to yes. Then, your back to both Asterisk servers being completely taken out of the loop again! reinvites only remove the server from the AUDIO PATH. Signaling is still going thru Asterisk no matter what happens. *nod* I know. :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi and reinvites...
On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote: That's all fine and good until it becomes the receiving phone, and the other phone (as an originator) also has canreinvite set to yes. Then, your back to both Asterisk servers being completely taken out of the loop again! While I haven't taken the time to actually try this, I might suggest that you could set up separate user and peer sections in sip.conf, so that you can handle inbound calls differently that outbound calls. -Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
