Ok, we're trying to use Asterisk as a PBX in our office. Our original plan was to use a Cisco 7960 with a 7914 attached. Short story is, no one updated chan_sccp in a long time and the 7914 is questionable at best anyway from what I've heard. We couldn't ever get chan_sccp to compile, I went to an older version of Asterisk and that broke some of our SIP devices. We tried using a couple soft panels listed on the Wiki, the only one easy enough to use was Asternic. And we found also that is buggy and doesn't function correctly with the new Asterisk. Any Suggestions? I was looking at the SNOM 220 (w/ op panel). We're mainly after the line status of course more than anything, wasn't sure if it could do that, SIP and all.. Matt
What was wrong with Asternic, I use it and it works fine. On Wed, 12 Jan 2005 08:07:11 -0600, Matt Schulte <mschulte@netlogic.net> wrote:> Ok, we're trying to use Asterisk as a PBX in our office. Our original > plan was to use a Cisco 7960 with a 7914 attached. Short story is, no > one updated chan_sccp in a long time and the 7914 is questionable at > best anyway from what I've heard. We couldn't ever get chan_sccp to > compile, I went to an older version of Asterisk and that broke some of > our SIP devices. We tried using a couple soft panels listed on the Wiki, > the only one easy enough to use was Asternic. And we found also that is > buggy and doesn't function correctly with the new Asterisk. Any > Suggestions? > > I was looking at the SNOM 220 (w/ op panel). We're mainly after the line > status of course more than anything, wasn't sure if it could do that, > SIP and all.. > > Matt > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users >
On Wed, Jan 12, 2005 at 08:07:11AM -0600, Matt Schulte arranged a set of bits into the following:> Ok, we're trying to use Asterisk as a PBX in our office. Our original > plan was to use a Cisco 7960 with a 7914 attached. Short story is, no > one updated chan_sccp in a long time and the 7914 is questionable at > best anyway from what I've heard. We couldn't ever get chan_sccp to > compile, I went to an older version of Asterisk and that broke some of > our SIP devices. We tried using a couple soft panels listed on the Wiki, > the only one easy enough to use was Asternic. And we found also that is > buggy and doesn't function correctly with the new Asterisk. Any > Suggestions?What was your problem with chan_sccp? There's only one small issue I know of in the code (already fixed, I just haven't committed it to CVS). Although the biggest issue with using it would be that chan_sccp doesn't yet have hint support (it's forthcoming once I get my new phone delivered this week). Thanks, Julien Goodwin chan_sccp developer -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 232 bytes Desc: not available Url : lists.digium.com/pipermail/asterisk-users/attachments/20050112/2313f864/attachment.pgp
on 'make' >> chan_sccp.c: In function `load_module': chan_sccp.c:653: warning: passing arg 4 of `ast_channel_register_ex' from incompatible pointer type Now compiling .... sccp_actions.c 743 lines Now compiling .... sccp_channel.c 279 lines sccp_channel.c: In function `sccp_channel_send_callinfo': sccp_channel.c:48: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' make: *** [.tmp/sccp_channel.o] Error 1 I would have posted in the sccp bugtracker but it looks like no one has used it in months. This is from cvs compile. We tried using stable ast (ie 1.0.2) but it broke our SIP (digest) authentication going to SER for some reason. Thanks, Matt -----Original Message----- From: Julien Goodwin [mailto:asterisk-lists@studio442.com.au] What was your problem with chan_sccp? There's only one small issue I know of in the code (already fixed, I just haven't committed it to CVS). Although the biggest issue with using it would be that chan_sccp doesn't yet have hint support (it's forthcoming once I get my new phone delivered this week). Thanks, Julien Goodwin chan_sccp developer
The problem we're having is transfers don't seem to work? ie: when someone calls inbound, you drag and drop the call on the extension you'd like and it just bridges the 2 phones together instead of transfering the call? Maybe this was intentional or maybe I'm just doing something wrong? Other than that the panel seems to work great. Matt -----Original Message----- From: C F [mailto:shmaltz@gmail.com] Sent: Wednesday, January 12, 2005 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Operator Panels? What was wrong with Asternic, I use it and it works fine. On Wed, 12 Jan 2005 08:07:11 -0600, Matt Schulte <mschulte@netlogic.net> wrote:> Ok, we're trying to use Asterisk as a PBX in our office. Our original > plan was to use a Cisco 7960 with a 7914 attached. Short story is, no > one updated chan_sccp in a long time and the 7914 is questionable at > best anyway from what I've heard. We couldn't ever get chan_sccp to > compile, I went to an older version of Asterisk and that broke some of> our SIP devices. We tried using a couple soft panels listed on the > Wiki, the only one easy enough to use was Asternic. And we found also > that is buggy and doesn't function correctly with the new Asterisk. > Any Suggestions? > > I was looking at the SNOM 220 (w/ op panel). We're mainly after the > line status of course more than anything, wasn't sure if it could do > that, SIP and all.. > > Matt > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users
Ok, I think I somewhat understand the concept now that I think about it. You only want to xfer your trunks correct? In other words Zaptel to user. We also have IAX trunks for outbound that support multiple lines. Creating one instance the the IAX trunk won't work because it can only show one call. I was thinking a solution may be to create an instance for several trunks ie: IAX2/blah/1 IAX2/blah/2 .. so forth The problem with that is 1) the trunk numbers can be random, or at least "stack" and eventually reset back down to one? I couldn't find a way to force these range of numbers, the wiki may be outdated though. Anyone have any thoughts on this? Matt -----Original Message----- From: Matt Schulte Sent: Wednesday, January 19, 2005 8:12 AM To: C F; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Operator Panels? The problem we're having is transfers don't seem to work? ie: when someone calls inbound, you drag and drop the call on the extension you'd like and it just bridges the 2 phones together instead of transfering the call? Maybe this was intentional or maybe I'm just doing something wrong? Other than that the panel seems to work great. Matt -----Original Message----- From: C F [mailto:shmaltz@gmail.com] Sent: Wednesday, January 12, 2005 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Operator Panels? What was wrong with Asternic, I use it and it works fine. On Wed, 12 Jan 2005 08:07:11 -0600, Matt Schulte <mschulte@netlogic.net> wrote:> Ok, we're trying to use Asterisk as a PBX in our office. Our original > plan was to use a Cisco 7960 with a 7914 attached. Short story is, no > one updated chan_sccp in a long time and the 7914 is questionable at > best anyway from what I've heard. We couldn't ever get chan_sccp to > compile, I went to an older version of Asterisk and that broke some of> our SIP devices. We tried using a couple soft panels listed on the > Wiki, the only one easy enough to use was Asternic. And we found also > that is buggy and doesn't function correctly with the new Asterisk. > Any Suggestions? > > I was looking at the SNOM 220 (w/ op panel). We're mainly after the > line status of course more than anything, wasn't sure if it could do > that, SIP and all.. > > Matt > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users
It's called asternic, asternic.org .. The client is based on flash which connects to a perl daemon on the server. It uses the manager (manager.conf) interface to determine extension status. Pretty neat :-) Matt -----Original Message----- From: David John Walsh [mailto:davidjwalsh@mac.com] Sent: Thursday, January 20, 2005 12:22 AM To: Nicol?s Gudi?o; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Operator Panels? On 20 Jan 2005, at 03:06, Nicol?s Gudi?o wrote:> Hello, > >> The problem we're having is transfers don't seem to work? ie: when >> someone calls inbound, you drag and drop the call on the extension >> you'd like and it just bridges the 2 phones together instead of >> transfering the call? Maybe this was intentional or maybe I'm just >> doing something wrong? Other than that the panel seems to work great. > > You can set reverse_transfer to 1 in op_server.cfg and it will > transfer the other leg of the call (Ex: if you drag phone A to phone > B, it will transfer the other leg of phone A (maybe an iax trunk or > whatever) to B, instead of dropping the trunk and bridging A with B. > >I am interested in the product that is being described here, but have only recently joined the discussion list. I guess my question is in 2 parts : a) what is the product / area of asterisk that is being refered to within this email b) is there an archive of messages for this reflector that I can browse before posting questions? Kind regards David _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users
I couldn't find this option, I'm running the latest "stable" there is an unstable version, is it in that one? -----Original Message----- From: Nicol?s Gudi?o [mailto:asternic@gmail.com] Sent: Wednesday, January 19, 2005 9:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Operator Panels? Hello,> The problem we're having is transfers don't seem to work? ie: when > someone calls inbound, you drag and drop the call on the extension > you'd like and it just bridges the 2 phones together instead of > transfering the call? Maybe this was intentional or maybe I'm just > doing something wrong? Other than that the panel seems to work great.You can set reverse_transfer to 1 in op_server.cfg and it will transfer the other leg of the call (Ex: if you drag phone A to phone B, it will transfer the other leg of phone A (maybe an iax trunk or whatever) to B, instead of dropping the trunk and bridging A with B. -- Nicol?s Gudi?o Buenos Aires - Argentina _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users