Re: [asterisk-users] Cisco 7961G + 7914 Expansion Module
On Tue, 2007-06-05 at 12:26 -0500, Jason Parker wrote: > - "Eric Lubow" <[EMAIL PROTECTED]> wrote: > > All, > > > >Since I have now (at least partially) got my 7961G phones working > > with Asterisk, I have temporarily moved on to try to get the > > expansion > > modules working. There doesn't seem to be much in the way of > > documentation here either. Does anyone have this combination working > > (or any 79X1) here? > > > >My goal is ultimately to do the monitoring approach. I have > > Google'd > > around, but come up with little. I want the office manager to be > > able > > to see when the user is on the phone or when the phone is ringing (or > > even be able to pick it up). > > > >I am looking to keep this operation within the SIP space, but if I > > have to go with SCCP to get this module working (if there is a way to > > make the phone treat the lines as protocol independent). Does anyone > > have any suggestions (or examples) as to how to accomplish this? > > Thanks. > > > > Eric > Last I knew, the expansion modules only worked with Skinny firmware. There > is support for it in chan_skinny in 1.4, but it's mostly useless, as they can > only be line appearances. svn trunk has support for speeddials/hints though. Does that mean that if I have the phone working with SIP that I can still use those line appearances for informational purposes? Is there an example or a HOWTO somewhere to use as a guide? Thanks. Eric -- Eric Lubow LinkExperts, Inc. Systems Administrator e: [EMAIL PROTECTED] w: www.linkexperts.com ___ --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] Cisco 7961G + 7914 Expansion Module
- "Eric Lubow" <[EMAIL PROTECTED]> wrote: > All, > >Since I have now (at least partially) got my 7961G phones working > with Asterisk, I have temporarily moved on to try to get the > expansion > modules working. There doesn't seem to be much in the way of > documentation here either. Does anyone have this combination working > (or any 79X1) here? > >My goal is ultimately to do the monitoring approach. I have > Google'd > around, but come up with little. I want the office manager to be > able > to see when the user is on the phone or when the phone is ringing (or > even be able to pick it up). > >I am looking to keep this operation within the SIP space, but if I > have to go with SCCP to get this module working (if there is a way to > make the phone treat the lines as protocol independent). Does anyone > have any suggestions (or examples) as to how to accomplish this? > Thanks. > > Eric > > -- > Eric Lubow > LinkExperts, Inc. > Systems Administrator > e: [EMAIL PROTECTED] > w: www.linkexperts.com Last I knew, the expansion modules only worked with Skinny firmware. There is support for it in chan_skinny in 1.4, but it's mostly useless, as they can only be line appearances. svn trunk has support for speeddials/hints though. -- Jason Parker Digium ___ --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] Cisco 7961G
here is my minimalistic .cnf.xml, that works for my 7941 SIP admin *** D-M-Y Central Europe Standard/Daylight Time 2000 5060 5061 192.168.0.100 true false g711a 0 Asterisk 9 SIP 961 192.168.0.100 961 PJ7961 961 *** 8299 21 Echo test 959 DRdialplan.xml *** SIP41.8-2-1S 1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37 Eric Lubow wrote: It sounds like you are telling me that it is likely a firmware issue and not an Asterisk issue. Would it be possible for someone to provide me with a copy of your SEP.cnf.xml file and whatever other files the phone uses so I can ensure that its not something else? Thanks. Eric On Fri, 2007-06-01 at 15:07 -0500, Greg Oliver wrote: On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote: we are using 7941 with sip v8.2(2)SR3, it working quite well ;-) Eric Lubow wrote: All, I am having a lot of trouble with the Cisco 7961G phones. I have managed to get them up and running with Asterisk to the point where I can get incoming calls and make outgoing calls. The problem is when I make outgoing calls or extension to extension calls, the calls die after 20 seconds. I have google'd around and came up with little that is of help. The firmware version I am using on the phone is 8.0.4SR1. I have tried tcpdumping the conversation and I see that the phone doesn't send the SIP/SDP ACK packet back to the remote end. Sometimes it does, but that's a rarity. There doesn't seem to be any rhyme or reason as to when it will send the SIP/SDP ACK. All I see is the following before the phone hangs up at 20 seconds (201 is the phone and 205 is the Asterisk Box): 10.230103 192.168.0.205 -> 192.168.0.201 SIP/SDP Status: 200 OK, with session description Is there a newer version of the firmware that fixes this? Is there a setting in Asterisk that can fix this? Any help is greatly appreciated. Thanks. Eric Anything older than 8.0.4SR2 is asking for grief. You cannot even download older from Cisco's website anymore. Those were their CallManager "transitional" loads from SCCP -> SIP that were riddled with bugs. -Greg ___ --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] Cisco 7961G
It sounds like you are telling me that it is likely a firmware issue and not an Asterisk issue. Would it be possible for someone to provide me with a copy of your SEP.cnf.xml file and whatever other files the phone uses so I can ensure that its not something else? Thanks. Eric On Fri, 2007-06-01 at 15:07 -0500, Greg Oliver wrote: > On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote: > > we are using 7941 with sip v8.2(2)SR3, it working quite well ;-) > > > > > > Eric Lubow wrote: > > > All, > > > > > >I am having a lot of trouble with the Cisco 7961G phones. I have > > > managed to get them up and running with Asterisk to the point where I > > > can get incoming calls and make outgoing calls. The problem is when I > > > make outgoing calls or extension to extension calls, the calls die after > > > 20 seconds. I have google'd around and came up with little that is of > > > help. The firmware version I am using on the phone is 8.0.4SR1. > > > > > >I have tried tcpdumping the conversation and I see that the phone > > > doesn't send the SIP/SDP ACK packet back to the remote end. Sometimes > > > it does, but that's a rarity. There doesn't seem to be any rhyme or > > > reason as to when it will send the SIP/SDP ACK. All I see is the > > > following before the phone hangs up at 20 seconds (201 is the phone and > > > 205 is the Asterisk Box): > > > > > > 10.230103 192.168.0.205 -> 192.168.0.201 SIP/SDP Status: 200 OK, with > > > session description > > > > > >Is there a newer version of the firmware that fixes this? Is there a > > > setting in Asterisk that can fix this? Any help is greatly appreciated. > > > Thanks. > > > > > > Eric > > > > Anything older than 8.0.4SR2 is asking for grief. You cannot even > download older from Cisco's website anymore. Those were their > CallManager "transitional" loads from SCCP -> SIP that were riddled with > bugs. > > -Greg > > ___ > --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 -- Eric Lubow LinkExperts, 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] Cisco 7961G
On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote: > we are using 7941 with sip v8.2(2)SR3, it working quite well ;-) > > > Eric Lubow wrote: > > All, > > > >I am having a lot of trouble with the Cisco 7961G phones. I have > > managed to get them up and running with Asterisk to the point where I > > can get incoming calls and make outgoing calls. The problem is when I > > make outgoing calls or extension to extension calls, the calls die after > > 20 seconds. I have google'd around and came up with little that is of > > help. The firmware version I am using on the phone is 8.0.4SR1. > > > >I have tried tcpdumping the conversation and I see that the phone > > doesn't send the SIP/SDP ACK packet back to the remote end. Sometimes > > it does, but that's a rarity. There doesn't seem to be any rhyme or > > reason as to when it will send the SIP/SDP ACK. All I see is the > > following before the phone hangs up at 20 seconds (201 is the phone and > > 205 is the Asterisk Box): > > > > 10.230103 192.168.0.205 -> 192.168.0.201 SIP/SDP Status: 200 OK, with > > session description > > > >Is there a newer version of the firmware that fixes this? Is there a > > setting in Asterisk that can fix this? Any help is greatly appreciated. > > Thanks. > > > > Eric > > Anything older than 8.0.4SR2 is asking for grief. You cannot even download older from Cisco's website anymore. Those were their CallManager "transitional" loads from SCCP -> SIP that were riddled with bugs. -Greg ___ --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] Cisco 7961G
we are using 7941 with sip v8.2(2)SR3, it working quite well ;-) Eric Lubow wrote: All, I am having a lot of trouble with the Cisco 7961G phones. I have managed to get them up and running with Asterisk to the point where I can get incoming calls and make outgoing calls. The problem is when I make outgoing calls or extension to extension calls, the calls die after 20 seconds. I have google'd around and came up with little that is of help. The firmware version I am using on the phone is 8.0.4SR1. I have tried tcpdumping the conversation and I see that the phone doesn't send the SIP/SDP ACK packet back to the remote end. Sometimes it does, but that's a rarity. There doesn't seem to be any rhyme or reason as to when it will send the SIP/SDP ACK. All I see is the following before the phone hangs up at 20 seconds (201 is the phone and 205 is the Asterisk Box): 10.230103 192.168.0.205 -> 192.168.0.201 SIP/SDP Status: 200 OK, with session description Is there a newer version of the firmware that fixes this? Is there a setting in Asterisk that can fix this? Any help is greatly appreciated. Thanks. Eric ___ --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
