Re: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP)
After reading my response I didn't make it all that obvious that I was also saying that the Cisco's don't support hints. Chalk it up to a long day. --J. Aaron Daniel wrote: Speaking as someone who did an installation with quite a few Cisco phones running sip, they do not support hinting at all with the sip firmware. The latest version of the Cisco phones (79x1) are primarily sip phones, however with the CCM specific extensions, they tend to be fairly difficult to get to work with asterisk. I wouldn't recommend banging your head on the wall trying, took a while for the bump on my head to go away. :) Anyone trying to get this to work would be better off getting a phone that does support hinting, and has plenty of documentation on how to do it (Polycom, Aastra, SNOM, etc). Jason Howk wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I run the phone with sip firmware so I can confirm it does. ;) Actually the "G" means global and replaces the actual text on the buttons with icons instead. The gigabit interfaces come on the later - -GE models. My question was more directed to if anyone has gotten SIP hints to work on the older 7960s at all. Looks like I might just have to give the new snom 370 a try... - --J. On Apr 25, 2007, at 7:59 PM, Brad Sumrall wrote: I am very confident the 7960G has a sip load. I know for sure the regular 7960 does and the G just means gigabit interface. The 7970 was the only one that didn't because of all the color interface/touch screen, and Cisco was still pushing call manager big time, so skinny was the only load available. If you log into cisco.com, they have it under software. Sometimes people post it on the internet. Asterisk is supposed to be more skinny friendly these days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Howk Sent: Wednesday, April 25, 2007 7:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP) From reading the SLA docs, SIP hints are use to get the lights on the phone to show the "correct state". I was under the impression that the SIP firmware on the 7960's didn't support the SIP hints properly (or at all), which means that SLA won't work properly on a 7960. If anyone has gotten this to work, I'd like to hear about it. --Jason. John C. Wolosuk Jr. wrote: Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should behave. It seems apparent that the phones should not register with asterisk, otherwise all the phones will try to register to be THE phone for a given extension. should these lines be built like a trunk/peer? if I could be an example of how lines for SLA should look in sip.conf, that would be helpful. Also I'm somewhat annoyed that I have to compile zaptel drivers that I don't use in order to compile the app_meetme.so module so I can have the SLA functions available to the dialplan... Any feedback is greatly appreciated! ___ --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 ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFGMD8nyocPzc/H1dsRAtlVAJ4gLjMENCyW2wDFMhxMRO6eIX76yQCdESBt G83ykWxG1EWcxLNqZfyp5ME= =RVLT -END PGP SIGNATURE- ___ --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 ___ --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] SLA Appearance between 2 Cisco 7960's (SIP)
Speaking as someone who did an installation with quite a few Cisco phones running sip, they do not support hinting at all with the sip firmware. The latest version of the Cisco phones (79x1) are primarily sip phones, however with the CCM specific extensions, they tend to be fairly difficult to get to work with asterisk. I wouldn't recommend banging your head on the wall trying, took a while for the bump on my head to go away. :) Anyone trying to get this to work would be better off getting a phone that does support hinting, and has plenty of documentation on how to do it (Polycom, Aastra, SNOM, etc). Jason Howk wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I run the phone with sip firmware so I can confirm it does. ;) Actually the "G" means global and replaces the actual text on the buttons with icons instead. The gigabit interfaces come on the later - -GE models. My question was more directed to if anyone has gotten SIP hints to work on the older 7960s at all. Looks like I might just have to give the new snom 370 a try... - --J. On Apr 25, 2007, at 7:59 PM, Brad Sumrall wrote: I am very confident the 7960G has a sip load. I know for sure the regular 7960 does and the G just means gigabit interface. The 7970 was the only one that didn't because of all the color interface/touch screen, and Cisco was still pushing call manager big time, so skinny was the only load available. If you log into cisco.com, they have it under software. Sometimes people post it on the internet. Asterisk is supposed to be more skinny friendly these days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Howk Sent: Wednesday, April 25, 2007 7:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP) From reading the SLA docs, SIP hints are use to get the lights on the phone to show the "correct state". I was under the impression that the SIP firmware on the 7960's didn't support the SIP hints properly (or at all), which means that SLA won't work properly on a 7960. If anyone has gotten this to work, I'd like to hear about it. --Jason. John C. Wolosuk Jr. wrote: Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should behave. It seems apparent that the phones should not register with asterisk, otherwise all the phones will try to register to be THE phone for a given extension. should these lines be built like a trunk/peer? if I could be an example of how lines for SLA should look in sip.conf, that would be helpful. Also I'm somewhat annoyed that I have to compile zaptel drivers that I don't use in order to compile the app_meetme.so module so I can have the SLA functions available to the dialplan... Any feedback is greatly appreciated! ___ --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 ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFGMD8nyocPzc/H1dsRAtlVAJ4gLjMENCyW2wDFMhxMRO6eIX76yQCdESBt G83ykWxG1EWcxLNqZfyp5ME= =RVLT -END PGP SIGNATURE- ___ --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 -- Aaron Daniel Community Relations Specialist [EMAIL PROTECTED] (256) 428-6010 ___ --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] SLA Appearance between 2 Cisco 7960's (SIP)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I run the phone with sip firmware so I can confirm it does. ;) Actually the "G" means global and replaces the actual text on the buttons with icons instead. The gigabit interfaces come on the later - -GE models. My question was more directed to if anyone has gotten SIP hints to work on the older 7960s at all. Looks like I might just have to give the new snom 370 a try... - --J. On Apr 25, 2007, at 7:59 PM, Brad Sumrall wrote: I am very confident the 7960G has a sip load. I know for sure the regular 7960 does and the G just means gigabit interface. The 7970 was the only one that didn't because of all the color interface/touch screen, and Cisco was still pushing call manager big time, so skinny was the only load available. If you log into cisco.com, they have it under software. Sometimes people post it on the internet. Asterisk is supposed to be more skinny friendly these days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Howk Sent: Wednesday, April 25, 2007 7:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP) From reading the SLA docs, SIP hints are use to get the lights on the phone to show the "correct state". I was under the impression that the SIP firmware on the 7960's didn't support the SIP hints properly (or at all), which means that SLA won't work properly on a 7960. If anyone has gotten this to work, I'd like to hear about it. --Jason. John C. Wolosuk Jr. wrote: Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should behave. It seems apparent that the phones should not register with asterisk, otherwise all the phones will try to register to be THE phone for a given extension. should these lines be built like a trunk/peer? if I could be an example of how lines for SLA should look in sip.conf, that would be helpful. Also I'm somewhat annoyed that I have to compile zaptel drivers that I don't use in order to compile the app_meetme.so module so I can have the SLA functions available to the dialplan... Any feedback is greatly appreciated! ___ --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 ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFGMD8nyocPzc/H1dsRAtlVAJ4gLjMENCyW2wDFMhxMRO6eIX76yQCdESBt G83ykWxG1EWcxLNqZfyp5ME= =RVLT -END PGP SIGNATURE- ___ --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] SLA Appearance between 2 Cisco 7960's (SIP)
I am very confident the 7960G has a sip load. I know for sure the regular 7960 does and the G just means gigabit interface. The 7970 was the only one that didn't because of all the color interface/touch screen, and Cisco was still pushing call manager big time, so skinny was the only load available. If you log into cisco.com, they have it under software. Sometimes people post it on the internet. Asterisk is supposed to be more skinny friendly these days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Howk Sent: Wednesday, April 25, 2007 7:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP) >From reading the SLA docs, SIP hints are use to get the lights on the phone to show the "correct state". I was under the impression that the SIP firmware on the 7960's didn't support the SIP hints properly (or at all), which means that SLA won't work properly on a 7960. If anyone has gotten this to work, I'd like to hear about it. --Jason. John C. Wolosuk Jr. wrote: > Has anyone had any success with getting SLA going between 2 SIP phones? > (Particularly a set of Cisco 79xx's) The SLA document that comes with > the asterisk source is about as clear as mud. > > Does anyone have a working sip.conf, sla.conf, and extensions.conf that > I can use for reference? > > The part I'm most confused about is how to build the lines in sip.conf > and how the phones should behave. It seems apparent that the phones > should not register with asterisk, otherwise all the phones will try to > register to be THE phone for a given extension. should these lines be > built like a trunk/peer? if I could be an example of how lines for SLA > should look in sip.conf, that would be helpful. > > Also I'm somewhat annoyed that I have to compile zaptel drivers that I > don't use in order to compile the app_meetme.so module so I can have the > SLA functions available to the dialplan... > > Any feedback is greatly appreciated! > ___ --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 ___ --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] SLA Appearance between 2 Cisco 7960's (SIP)
>From reading the SLA docs, SIP hints are use to get the lights on the phone to show the "correct state". I was under the impression that the SIP firmware on the 7960's didn't support the SIP hints properly (or at all), which means that SLA won't work properly on a 7960. If anyone has gotten this to work, I'd like to hear about it. --Jason. John C. Wolosuk Jr. wrote: > Has anyone had any success with getting SLA going between 2 SIP phones? > (Particularly a set of Cisco 79xx's) The SLA document that comes with > the asterisk source is about as clear as mud. > > Does anyone have a working sip.conf, sla.conf, and extensions.conf that > I can use for reference? > > The part I'm most confused about is how to build the lines in sip.conf > and how the phones should behave. It seems apparent that the phones > should not register with asterisk, otherwise all the phones will try to > register to be THE phone for a given extension. should these lines be > built like a trunk/peer? if I could be an example of how lines for SLA > should look in sip.conf, that would be helpful. > > Also I'm somewhat annoyed that I have to compile zaptel drivers that I > don't use in order to compile the app_meetme.so module so I can have the > SLA functions available to the dialplan... > > Any feedback is greatly appreciated! > ___ --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] SLA Appearance between 2 Cisco 7960's (SIP)
Russell Bryant wrote: John C. Wolosuk Jr. wrote: Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Mud, huh? I guess I should work on that at some point, then ... You say two phones. What do you intend to use on the trunk side? I assume you want a SIP trunk. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? sip.conf: This is configured just like any other SIP device. In your scenario of two SIP phones and one SIP trunk, sip.conf would contain three entries. For example: [station1] type=friend secret=station1 host=dynamic context=sla_stations dtmfmode=rfc2833 disallow=all allow=ulaw [station2] type=friend secret=station2 host=dynamic context=sla_stations dtmfmode=rfc2833 disallow=all allow=ulaw [providerA] type=friend secret=something host=providerA.com context=line1 dtmfmode=rfc2833 disallow=all allow=ulaw sla.conf: (From sla.pdf, page 7) Here you create a definition for a single line and two stations. [line1] type=trunk device=Local/[EMAIL PROTECTED] [station](!) type=station trunk=line1 [station1](station) device=SIP/station1 [station2](station) device=SIP/station2 extensions.conf: [line1] ; This is used for incoming calls from SIP/providerA because providerA ; has context=line1 in sip.conf. Incoming calls immediately go into the ; SLATrunk application. Then, the appropriate stations will start ; ringing. exten => s,1,SLATrunk(line1) [line1_outbound] ; This context is used by the SLA code. line1 in sla.conf was ; configured to use a device called Local/[EMAIL PROTECTED] ; That means that when someone presses the line button for line1, ; it will get connected to Disa. Disa will provide dialtone and ; allow the caller to dial any other extensions that live in this ; context. In this case, there is only one available pattern. When ; it gets dialed, the call goes out to SIP/providerA. exten => disa,1,Disa(no-password|line1_outbound) exten => _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) [sla_stations] ; These extensions are called by the stations . ; This extension should be called when the the phone for ; SIP/station1 is taken off hook without pressing a line button. exten => station1,1,SLAStation(station1) ; This extension should be called when the user presses the ; line1 key on the phone. exten => station1_line1,1,SLAStation(station1_line1) ; The line1 key on the phone for station1 should be configured ; to subscribe to the state of the extension "station1_line1". ; This will allow Asterisk to control the light to make it turn ; on, off, or blink, as appropriate. exten => station1_line1,hint,SLA:station1_line1 exten => station2,1,SLAStation(station2) exten => station2_line1,hint,SLA:station2_line1 exten => station2_line1,1,SLAStation(station2_line1) The part I'm most confused about is how to build the lines in sip.conf and how the phones should behave. It seems apparent that the phones should not register with asterisk, otherwise all the phones will try to register to be THE phone for a given extension. should these lines be built like a trunk/peer? if I could be an example of how lines for SLA should look in sip.conf, that would be helpful. Actually, the phones *do* register to Asterisk. But, the line appearance buttons themselves are not registrations to Asterisk. They are simply subscribers to the state of extensions. You set these up just like you would for any other hint in Asterisk. Just an FYI, Cisco phones running SIP do *not* do shared line appearances, on *any* system. -- Aaron Daniel Community Relations Specialist [EMAIL PROTECTED] (256) 428-6010 ___ --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] SLA Appearance between 2 Cisco 7960's (SIP)
John C. Wolosuk Jr. wrote: Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Mud, huh? I guess I should work on that at some point, then ... You say two phones. What do you intend to use on the trunk side? I assume you want a SIP trunk. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? sip.conf: This is configured just like any other SIP device. In your scenario of two SIP phones and one SIP trunk, sip.conf would contain three entries. For example: [station1] type=friend secret=station1 host=dynamic context=sla_stations dtmfmode=rfc2833 disallow=all allow=ulaw [station2] type=friend secret=station2 host=dynamic context=sla_stations dtmfmode=rfc2833 disallow=all allow=ulaw [providerA] type=friend secret=something host=providerA.com context=line1 dtmfmode=rfc2833 disallow=all allow=ulaw sla.conf: (From sla.pdf, page 7) Here you create a definition for a single line and two stations. [line1] type=trunk device=Local/[EMAIL PROTECTED] [station](!) type=station trunk=line1 [station1](station) device=SIP/station1 [station2](station) device=SIP/station2 extensions.conf: [line1] ; This is used for incoming calls from SIP/providerA because providerA ; has context=line1 in sip.conf. Incoming calls immediately go into the ; SLATrunk application. Then, the appropriate stations will start ; ringing. exten => s,1,SLATrunk(line1) [line1_outbound] ; This context is used by the SLA code. line1 in sla.conf was ; configured to use a device called Local/[EMAIL PROTECTED] ; That means that when someone presses the line button for line1, ; it will get connected to Disa. Disa will provide dialtone and ; allow the caller to dial any other extensions that live in this ; context. In this case, there is only one available pattern. When ; it gets dialed, the call goes out to SIP/providerA. exten => disa,1,Disa(no-password|line1_outbound) exten => _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) [sla_stations] ; These extensions are called by the stations . ; This extension should be called when the the phone for ; SIP/station1 is taken off hook without pressing a line button. exten => station1,1,SLAStation(station1) ; This extension should be called when the user presses the ; line1 key on the phone. exten => station1_line1,1,SLAStation(station1_line1) ; The line1 key on the phone for station1 should be configured ; to subscribe to the state of the extension "station1_line1". ; This will allow Asterisk to control the light to make it turn ; on, off, or blink, as appropriate. exten => station1_line1,hint,SLA:station1_line1 exten => station2,1,SLAStation(station2) exten => station2_line1,hint,SLA:station2_line1 exten => station2_line1,1,SLAStation(station2_line1) The part I'm most confused about is how to build the lines in sip.conf and how the phones should behave. It seems apparent that the phones should not register with asterisk, otherwise all the phones will try to register to be THE phone for a given extension. should these lines be built like a trunk/peer? if I could be an example of how lines for SLA should look in sip.conf, that would be helpful. Actually, the phones *do* register to Asterisk. But, the line appearance buttons themselves are not registrations to Asterisk. They are simply subscribers to the state of extensions. You set these up just like you would for any other hint in Asterisk. Also I'm somewhat annoyed that I have to compile zaptel drivers that I don't use in order to compile the app_meetme.so module so I can have the SLA functions available to the dialplan... It uses MeetMe internally, and MeetMe requires Zaptel. That's just the way it is. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] SLA Appearance between 2 Cisco 7960's (SIP)
The zaptel stuff compiles fine, just need to know how to properly configure SLA for the SIP world. Stephen Bosch wrote: John C. Wolosuk Jr. wrote: Also I'm somewhat annoyed that I have to compile zaptel drivers that I don't use in order to compile the app_meetme.so module so I can have the SLA functions available to the dialplan... If you're using SLA, you're using zaptel drivers, yes -- without the timing source from ztdummy your SLA won't work (that's really what SLA is -- a fancy Meetme conference). Are you having trouble compiling Zaptel? -Stephen- ___ --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 ___ --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] SLA Appearance between 2 Cisco 7960's (SIP)
John C. Wolosuk Jr. wrote: > Also I'm somewhat annoyed that I have to compile zaptel drivers that I > don't use in order to compile the app_meetme.so module so I can have the > SLA functions available to the dialplan... If you're using SLA, you're using zaptel drivers, yes -- without the timing source from ztdummy your SLA won't work (that's really what SLA is -- a fancy Meetme conference). Are you having trouble compiling Zaptel? -Stephen- ___ --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
