Re: [Asterisk-Users] Asterisk Passthrough
On Tue, Mar 02, 2004 at 02:19:16PM -0500, Steve Creel wrote: [incoming] exten = 12345,1,Dial(ZAP/g2) ; Send incoming call for 12345 to the PBX FWIW, I've done something like this and it was absolutely wonderful. We were actually running new phones and putting them in parallel with the existing system over the same phone lines (they ran 4-pair UTP to each phone jack, so we just stole the outer pair and bought some magic adapters to pull out the second line). You can imagine the surprise when both phones worked simultaneously. It was even more surprising when we were training people on the new phones by having them dial an outside line and then use the new dial codes for voicemail and such. It went over VERY well. How I can forward a call? It's simply an extension.conf rule? Yes. Most people miss this. Use the Dial application (as the example shows). Dial is used for outside lines and such, but it is *'s fundamental way to make one channel dial to another. Virtually every situation where you are forwarding something (say Zap to SIP, SIP to IAX, TDM to TDMoe) you end up using a Dial. When I make the forward in this way (with extension.conf rule) asterisk make some work or is a simple passthrough from interfaces? Yes, it's some switching/callsetup work, but no codec translation, which is by far your biggest CPU consumer. It acts as a simple passthru for the CHANNELS. That is, what comes in on an individual channel goes out on another. Mapping the whole T1 would be another story (it can be done, I had to once). DACS works well for that but Asterisk can't get at the calls. I recommend the above. The only time it doesn't work well is when people want to do something with line 5. I had a situation where certain lines couldn't dial long distance. Since the above would dynamically choose a line, it would cause unexpected problems because the old PBX's line X was no longer actually the same T1 channel on the outside. I need that calls from PRI to PRI don't load the computer. I want to use all CPU to (future) SIP calls. Once the call is linked, all the load is on the Zaptel board. That is REALLY handy. I can't tell you how surprised some of my customers get when I have three machines switching 300 lines with like 5% or so load a piece. Feel free to e-mail me or jabber me (same as my e-mail address) if you have problems. I love to help set things like this up--especially in an more casual setting (you never get to have FUN with people's businesses). Jayson Vantuyl ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk Passthrough
On Tue, 2 Mar 2004, Emanuele Laface wrote: My actual pbx is connected to the external world with 2 PRI interface, my idea is to insert asterisk in the middle, I want disconnect the two PRI, connect them to the asterisk and connect the asterisk with old pbx with a cross cable. So, at the first step, my asterisk is simple a passthrough, but in the future I can change smoothly all my office phone and finally I can disconnect the old pbx. You've got two options here - you can use dacs in /etc/zaptel.conf to literally just cross-connect the PRIs. The two telco PRIs would come in on two of your ports, and would turn around and go back out on the other two. This happens in the zaptel module and doesn't make it up into asterisk. Your other option is to terminate the two PRIs into asterisk, and use asterisk to provide two PRIs into your PBX. This gives you access to the actual call routing. Ok, I'm at the first step, I have two problem: - First problem: what is the configuration of asterisk for passthrough? I have a good knowledge about SIP, IAX, and asterisk in general, I have build a working configuration with SIP phones (Grandstream Budget One) and asterisk with a PRI, but I don't know how I can configure the zaptel card for passthrough. - Second problem (is not a real problem): where I can find the diagram for a cross cable for PRI-to-PRI connection. If you're looking for a cable to go from the TE410P to your existing phone switch, you need a T1 crossover. Jared Smith has a good chart: http://www.jaredsmith.net/misc/cables/ Good luck, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk Passthrough
You've got two options here - you can use dacs in /etc/zaptel.conf to literally just cross-connect the PRIs. The two telco PRIs would come in on two of your ports, and would turn around and go back out on the other two. This happens in the zaptel module and doesn't make it up into asterisk. Can you elaborate on this? I have no mention of the term 'dacs' in /etc/zaptel.conf. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk Passthrough
On Tue, 2 Mar 2004, Steve Creel wrote: Your other option is to terminate the two PRIs into asterisk, and use asterisk to provide two PRIs into your PBX. This gives you access to the actual call routing. Ok, my problem is exactly how I can do that? How can I say to asterisk this port is connected to the world and the other is connected to my office telephon switch (this is my main problem, I see something about groups but I'm not sure about the right configuration...)? How I can forward a call? It's simply an extension.conf rule? When I make the forward in this way (with extension.conf rule) asterisk make some work or is a simple passthrough from interfaces? I need that calls from PRI to PRI don't load the computer. I want to use all CPU to (future) SIP calls. Thank you for your reply. Ciao Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk Passthrough
On Tue, 2 Mar 2004, Andrew Kohlsmith wrote: You've got two options here - you can use dacs in /etc/zaptel.conf to literally just cross-connect the PRIs. The two telco PRIs would come in on two of your ports, and would turn around and go back out on the other two. This happens in the zaptel module and doesn't make it up into asterisk. Can you elaborate on this? I have no mention of the term 'dacs' in /etc/zaptel.conf. According to asterisk-cvs, on October 30, 2003, dacs support was added: Add DACS functionality to zaptel for cross connecting channels zaptel.conf.sample is appropriately documented: dacs The zaptel driver cross connects the channels starting at the channel number listed at the end, after a colon If I wanted to cross connect the first span to the second: dacs = 1-24:25 If I want to cross connect just channel 3 to channel 27: dacs = 3:27 I imagine (though haven't tried it), you can use: dacs = 1,3-5:25 to take channels 1,3,4,5 and put them on 25,26,27,28 One note: you can only use dacs on T1/E1 spans, not the pci fxs/fxo cards. Hope that helps... Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk Passthrough
On Tue, 2 Mar 2004, Emanuele Laface wrote: On Tue, 2 Mar 2004, Steve Creel wrote: Your other option is to terminate the two PRIs into asterisk, and use asterisk to provide two PRIs into your PBX. This gives you access to the actual call routing. Ok, my problem is exactly how I can do that? How can I say to asterisk this port is connected to the world and the other is connected to my office telephon switch (this is my main problem, I see something about groups but I'm not sure about the right configuration...)? Don't try to map port to port - you're making your problem more complex than it needs to be. Let asterisk do some call routing for you. You've got an incoming call with dialed number identification. Write an asterisk extension rule to handle it... Should asterisk send it out on a specific channel? Should it be sent out to one channel out of a group? For example, say your incoming number is 12345. You've connected the telco PRIs to spans 1 and 2, and your PRIs to the existing PBX are spans 3 and 4. The telco channels are all in group 1, the channels to the PBX are in group 2. [incoming] exten = 12345,1,Dial(ZAP/g2) ; Send incoming call for 12345 to the PBX Now you have asterisk switching the call instead of cross connecting the ports. How I can forward a call? It's simply an extension.conf rule? Yes. When I make the forward in this way (with extension.conf rule) asterisk make some work or is a simple passthrough from interfaces? Yes, it's some switching/callsetup work, but no codec translation, which is by far your biggest CPU consumer. I need that calls from PRI to PRI don't load the computer. I want to use all CPU to (future) SIP calls. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
