Re: [Asterisk-Users] Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)
On Wed, Mar 02, 2005 at 12:42:11PM -0800, Don Murray wrote: Hmmm... I have this aweful feeling that I'm choosing the exact wrong time to ask a newbie question :) Oh well, here it goes. The quick question is : How do I dial an extension? (answer is probably - you don't in which case:) How do I dial my asterisk box? - I have no outside line, I just want to start testing things like voicemail internally. snip No stupid question here, you've obviously done your homework. You should look up breifly in the docs about contexts and extensions. According to the context line in your sip.conf, when those phones dial, they will be in context sip. Go to your extensions.conf and check you have something defined there. What you'd expect is something like: [sip] exten = 6000,1,Dial(SIP/175polycom) exten = 6001,1,Dial(SIP/175polycom) exten = 6010,1,Goto(demo,s,1) ; Just for fun... Then they can use 6000 and 6001 to call themselves and eachother. This should be enough to get you started. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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] Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)
Hey Don Have you set up lines in extensions.conf for your two phones? You want something like: exten = 175,1,Dial(SIP/175polycom) exten = 176,1,Dial(SIP/176polycom) Ideally you'd actually want those lines to point to a macro that handled voicemail on busy/no reply etc, but that's enough to get you going. On the Polycoms, you might want to edit the dial plan so it doesn't require a timeout or you pressing # (or the Send softkey) after the 3 digits, but that's not vital. If you're still stuck, give me a shout and I'll go through it with you. Cheers Paul ___ 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] Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)
*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 176polycom 192.168.0.176 255.255.255.255 5060 Unmonitored 175polycom 192.168.0.175 255.255.255.255 5060 Unmonitored Added to sip.conf: [175polycom] type=friend host=192.168.0.175 defaultip=192.168.0.175 dtmfmode=inband mailbox=175 context=sip callerid=I am Don progressinband=no ;polycom's seem to have trouble with the default progressinband=never [176polycom] type=friend host=192.168.0.176 defaultip=192.168.0.176 dtmfmode=inband mailbox=176 context=sip callerid=I am a jerk progressinband=no ;polycom's seem to have trouble with the default progressinband=never Don, I would get rid of the number/name combo and use just a number. [175] type=friend host=192.168.0.175 defaultip=192.168.0.175 dtmfmode=inband mailbox=175 context=sip callerid=I am Don progressinband=no ;polycom's seem to have trouble with the default progressinband=never In extensions.conf in your [sip] context add exten = _17X,1,Macro(stdexten) exten = _17X,2,Hangup Regards Dave ___ 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] Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)
Hmmm... I have this aweful feeling that I'm choosing the exact wrong time to ask a newbie question :) Oh well, here it goes. The quick question is : How do I dial an extension? (answer is probably - you don't in which case:) How do I dial my asterisk box? - I have no outside line, I just want to start testing things like voicemail internally. The details: I am not connected to the outside world yet, I have a couple of phones in-house and I'm trying to set up an Asterisk internal office phone network just to get my head wrapped around the system. I have - my linux box set up - the phones ftp'ing their latest firmware and config files - I can call one phone from the other using the IP address (no asterisk required) - I have installed zaptel, libpri, asterisk, asterisk samples - I have added my 2 phones to the sip.conf file (see below) - I see the two phones if I do a sip show peers with the correct IP addresses - I've tried to set up the phones as described at http://www.csh.rit.edu/~adamf/IP500.html; In the QuickStart guide it says that the way to test things are working is to call extension 1000 to get an automated message. Clearly the phones can talk to each other, I just want to take the next step to see if they can talk to Asterisk. Yet I can find nothing about extensions in any of the Polycom documentation, phone buttons and menus, etc, and I am beginning to think that the concept of an extension is an analogue phone thing and just doesn't make sense for IP phones. Anyway, I would really appreciate someone stopping on the shoulder, here, and helping me drag myself out of the ditch so I can careen down the highway, obstructing other people's progress as a newbie should... any help would be much appreciated. I feel like I am suffering from a fundamental disconnect. I can read and somewhat understand the details of the documentation regarding dialplan etc, I just don't know where the on ramp is, i.e. how to even talk to Asterisk with a phone, with my current set up. The only modifications I did were to added my asterisk server IP into the sip.cfg for the Polycom ftp account and to add the below into my /etc/asterisk/sip.conf file. Aside from that I'm working with a straight out of the box asterisk make; make install; make samples. Thanks in advance, Don *CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 176polycom 192.168.0.176 255.255.255.255 5060 Unmonitored 175polycom 192.168.0.175 255.255.255.255 5060 Unmonitored Added to sip.conf: [175polycom] type=friend host=192.168.0.175 defaultip=192.168.0.175 dtmfmode=inband mailbox=175 context=sip callerid=I am Don progressinband=no ;polycom's seem to have trouble with the default progressinband=never [176polycom] type=friend host=192.168.0.176 defaultip=192.168.0.176 dtmfmode=inband mailbox=176 context=sip callerid=I am a jerk progressinband=no ;polycom's seem to have trouble with the default progressinband=never You're almost there...you have the phones set to the 'sip' context. Edit your extensions.conf file, create a new context called [sip] (if it isn't already there), and add a dial statement to reach each phone: [sip] exten = 175,Dial(SIP/175polycom) Exten = 176,Dial(SIP/176polycom) (There are plenty of dial modifiers you can use, but there's a basic wau to get started...now just dial 175 and 176 from each phone respecitively to reach the opposite... Marty ___ 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] Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)
Guys, thanks for the help... Reading what Paul and Marty recommended made me start to understand what that context= field in the phone definition really was. I changed the context= to context=demo and dialled 1000 and got the Asterisk demo working! Great! :) I then switched back to context=sip in the sip.conf, added in the extensions.conf a [sip] section as suggested by Martijin and could dial between the phones using the extensions, plus get the demo! So, anyway, I think I've found the on-ramp, thanks a lot! I'll review Noah's and David's posts for further tips to improve this base and then go back to the handbook. The complexity is still a little daunting but I have 3 months before I need to get an operational system up. Thanks again, Don ___ 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
