Re: [Asterisk-Users] Re: sipura 841 mass provisioning
On Mar 2, 2006, at 2:15 AM, Vahan Yerkanian wrote: Reboot once again and it picks up the new config. Two-step provisioning takes a couple of reboots to insure the device has reconfigured itself. Applies to 2100, 3000, 841 and 941 models. I've had good results on our 942 by setting the resync interval extremely short in the initial configuration file -- it doesn't seem to immediately follow the new config file path the way the Snoms do: Resync_Periodic ua=na 5 /Resync_Periodic Resync_Error_Retry_Delay ua=na 60 /Resync_Error_Retry_Delay Just set it back to normal in the device-specific config (which in our case is generated by a script, so the initial file only serves to forward the phone to the real config source). -- Joshua P. Dady http://www.indecisive.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
[Asterisk-Users] SPA-941/2 Monitoring
(now that I've remembered which address is subscribed to this list) Does anyone with one of these phones have any sort of presence working? I'm looking to monitor the DND state of the phones, if nothing else. Setting the SIP-B bit enables SUBSCRIBE/NOTIFY, but the dialog package is the only one I know of the phone implementing, and it doesn't fire when DND changes (I'm ignoring whether it should; I'm just saying it doesn't). -- Joshua P. Dady ___ --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] Re: SNOM extension lights programmable, eg. based on asterisk variable setting?
On Jun 6, 2005, at 4:39 PM, [EMAIL PROTECTED] wrote:
I want to manage this dialplan variable for each extension separately,
unfortunately this doesn't work:
**77,hint,DS/splat${CALLERIDNUM}
Do you have an idea for that?
Is there an easy place to patch it in asterisk 1.0.7 stable?
Will it be possible with CVS-head?
Asterisk doesn't interpret variables in hints. I don't know about
other branches, I'm on the stable branch myself. I generate
fragments of my sip.conf, voicemail.conf, and extensions.conf from a
script and then #include them; you could try something like this.
--
Joshua P. Dady
smime.p7s
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Re: [Asterisk-Users] SNOM extension lights programmable, eg. based on asterisk variable setting?
On Jun 4, 2005, at 4:52 AM, [EMAIL PROTECTED] wrote: I would like the SNOM extension light to permanently reflect the current toggle status of my application logic/asterisk DB variable. There's a phantom device in bristuff that can be used for this sort of thing. When you toggle the dialplan variable, you'd do: **77,n,Devstate(splat|2) And then set a hint: **77,hint,DS/splat Ultimately it stores device states in same database you're using, so you could switch on them directly rather than keeping a second set of variables and needing to keep them in sync. -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ 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] snom and hint priority
On Apr 17, 2005, at 12:23 AM, Lance Grover wrote: I have rebooted the phone and restarted asterisk after each change. Did you do it in that order? If so, that is probably a source of trouble (you should restart or reload asterisk before the phone boots, not after). -- Joshua P. Dady http://www.indecisive.com/ smime.p7s Description: S/MIME cryptographic signature ___ 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] snom and hint priority
(boy mail in this list piles up fast when I can't check it) On Apr 8, 2005, at 10:03 AM, Michael George wrote: - It appears that the extension used with the hint must be the same as the extension used to dial that channel. So if extension 22 will ring Zap/2, then exten = 22,hint,Zap/2 will work, but exten = 222,hint,Zap/2 will not. Why is that? The extension is how asterisk maps SIP URLs to chunks of your dialplan -- if you program a button on a snom to dest sip:[EMAIL PROTECTED], the phone will use that same URL for both dialing and subscribing to extension state. Unless you have a phone that lets you specify different URLs for dialing and subscribing to state, they have to match in asterisk. - If I am correct in the above, then there is no way for me to monitor a channel that is not an extension. As an example, I have a TDM400 with 3 FXS (Zap/1-3 on extensions 21-23) and 1 FXO (Zap/4) as well as a VoIP channel for dialing out. I can monitor the states of the extensions with extension entries like exten = 21,hint,Zap/1 but I cannot monitor the state of the FXO with exten = 0,hint,Zap/4 because 0 is not the extension of Zap/4. Indeed, Zap/4 has no extension. Is it not possible to monitor that line, then? There has to be a SIP URL for the phone to subscribe to -- if you put: exten = zap4,hint,Zap/4 in your extensions.conf (with no zap4,1,... entry) it wouldn't be dialable (although the phone would still try if you pushed it) but would have a valid SIP URL. -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ 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] Snom and Multiple calls
On Apr 3, 2005, at 3:37 PM, Philipp von Klitzing wrote: - quality of the handset and the speaker phone? Our primary issue with the handset on the 220 was that the hook was really easy to miss; the 360 is easy to hang up without making your caller think you dropped the phone several times trying to do so. We haven't done enough testing with the speakerphone yet to really say anything definitively. - how do the buttons feel? Much better than the 220. Also, the icons didn't really mean anything to me on the 220 (which I presume is because Snom is european and I'm not); the same icons are used on the 360, but there's a text label below the button. - are the line LEDs multi-colour ones? Not that I've noticed. As the 360 softphone is available to everyone the software part isn't _that_ interesting as I can play with it myself... :-) The 360 softphone would be more interesting if you could magnify it so you can actually see the buttons. :) -- Joshua P. Dady http://www.indecisive.com/ smime.p7s Description: S/MIME cryptographic signature ___ 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] Re: Snom and Multiple calls
Okay, after talking with Sven today, it turns out my problem description is wrong (I was combining to cases, one of which does work in the current firmware): - Multiple incoming calls (works already) - Incoming call while dialing (or waiting for answer of) outgoing call (doesn't) -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ 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
[Asterisk-Users] Snom and Multiple calls
I've got an issue on the snoms, and I'm wondering if anyone has some recent experience with it; I've contacted the one specific reference I found to it in the list archives, and the person in question didn't seem to find an answer (and snom doesn't appear to be finished moving their offices yet). On the snom (I've tested this on the 220 and 360), the phone will immediately reject any new INVITE that arrives with 486 BUSY HERE if there's already a call on the phone opening (i.e., either the phone is already ringing or you've dialed a call that hasn't been answered yet). If we were to give one of these phones to our receptionist, obviously, that wouldn't be acceptable. Is there a way to disable this behavior? -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ 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
[Asterisk-Users] Re: Snom and Multiple calls
On Apr 1, 2005, at 10:52 AM, Noah Miller wrote: I don't have a 220, and I haven't really tested the 360, but on our 190's I just register each line appearance to the same sip device, and multiple simultaneous calls automatically roll from line 1 to line 2 to line 3, etc. Are you using any CheckGroup/Setgroup statements, or outgoinglimit? I've got one registration on the phone with no restrictions in asterisk about the number of simultaneous calls. Call waiting is enabled on the phones. If you're talking to someone and a second call comes in, the phone gives normal call-waiting indications. If, on the other hand, you're dialing a call, waiting for the other side to answer or there's already one ringing call, the incoming call is immediately rejected by the phone (busy here). -- Joshua P. Dady http://www.indecisive.com/ smime.p7s Description: S/MIME cryptographic signature ___ 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
[Asterisk-Users] Re: Snom and Multiple calls
On Apr 1, 2005, at 12:20 PM, Noah Miller wrote: This is just the way that Snom phones work. They say they want this behavior to keep things simple. Here is an old message from this list (Christian works for Snom): http://lists.digium.com/pipermail/asterisk-users/2004-April/044273.html - Noah P.S. Just to reiterate the point: Snom - I'd prefer a second ringing call to ring to another line appearance rather than have the call get sent back as busy. Maybe there can be an option? Yeah, and we're probably not the only ones. The 360 is so far ahead of anything else we've evaluated in terms of requirements, it would suck for this to be the deal-breaker. -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ 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
[Asterisk-Users] Re: Snom and Multiple calls
On Apr 1, 2005, at 1:15 PM, Noah Miller wrote: The vast majority of our handsets are Polycoms. I know that they do this correctly (with a little help from CheckGroup/SetGroup and multiple SIP registrations). Of course, you can't get the nifty sidecar for the Polycoms like you can for the 220. You can get a sidecar for the Cisco 7960, though, and they can do multiple lines, correctly, too. The Cisco phones don't do presence in SIP mode or hot keypad (plus I'm not especially fond of the interface). The Polycom does presence, and very nearly does hot keypad, but I get more information about Polycom's plans from my cat than I do from Polycom, and that worries me. Every part of the GXP-2000 needs to weigh more (plus the firmware on the one we had a chance to eval was clearly not done cooking yet). Don't even mention Mitel. :) Nothing has the number of sidecar buttons we want. The Snom really is the closest on all fronts (at least it is now that we've got a 360 and it fixes the case design issues we had with the 220). -- Joshua P. Dady http://www.indecisive.com/ smime.p7s Description: S/MIME cryptographic signature ___ 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
[Asterisk-Users] Re: Snom and Multiple calls
On Apr 1, 2005, at 2:34 PM, Noah Miller wrote: Just so you know, I don't think anybody has gotten the Polycom presence features to work properly with Asterisk. Am I wrong, anybody? If we went that way, I was going to handle the presence directly (i.e., a separate python script that subscribes to the phones' presence to watch for the bits we're interested in changing). -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ 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
[Asterisk-Users] Hold Pickup
I'm working through my list of features people will expect, and Hold Pickup is at the top at the moment -- has anyone done any work on this? We've had some unpleasant experiences with call parking, and everyone seems to like the Hold Pickup model. If you don't know what I mean by Hold Pickup, it's sort of a reverse transfer; pick up the nearest phone and dial prefix12345 to pick up a call holding on ext. 12345. It looks like the closest to what I want (without changing Asterisk) would be Park followed by an AGI that pokes my manager-port client (which in turn would redirect the target extension's channel to...well, something) -- if the parking space was returned to the dialplan somehow (or if Park() didn't ignore its arguments). At the moment, I'm using the v1-0 branch and at this point it looks like our phones will be all SIP (i.e., madding chan_sip to make sendonly channels visible to the dialplan somehow isn't automatically out of the question). -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ 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] Polycom vs. Cisco IP Phones
On Mar 18, 2005, at 8:27 AM, Ben Ruset wrote: 3. They don't realy support their phones, unless there is a hardware problem. They don't support them with Asterisk, but if you don't tell them about it, they tend to be very good at working to resolve issues. You have to know what issues they consider to be related to the platform. In general, copper and plastic issues (i.e., the phone is in the wrong number of pieces) the direct customer support people can help you with. As soon as you start talking about configuration, though, don't bother trying to get any specifics out of them that aren't in the admin guide -- you'd be wasting your time (as have many on this list before you). -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ 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
[Asterisk-Users] Re: Snom190 intercom
As you can see from the SIP trace below (from the called phone), intercom=true is being appended to the To: header as per requirements. The intercom=true needs to be appended to the request URI, not to the header as a whole -- your To: header should be: To: sip:1011 at 192.168.10.150:2051;line=9avrmhew;intercom=true Mind you, I didn't get the phone to respond to the intercom=true until I added it on the request line as well, so the INVITE line of your request would be: INVITE sip:1011 at 192.168.10.150:2051;line=9avrmhew;intercom=true SIP/2.0 I'm on a Snom 220 with SIP 3.56t, and I'm stuck on the very next step of the process -- answering the phone's challenge to the INVITE request. The wiki indicates that the Snom needs to challenge with realm=snom, but even if I add snom into our internal DNS so that I can set the registrar to snom (that being the only way I can see to change what the phone uses as realm), it still rejects the digest response. Anyone have this working with recent loads of SIP that can shed any light on this? I've email'd snom a few days ago but have yet to get a response. According to their web page, they have a new office as of April 1, and I got a response to a support request (on this very issue) today saying that they'd likely not be able to respond until people are settled into the new offices, so you'll likely have to be patient with them. -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ 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
[Asterisk-Users] Re: Snom190 intercom
As you can see from the SIP trace below (from the called phone), intercom=true is being appended to the To: header as per requirements. The intercom=true needs to be appended to the request URI, not to the header as a whole -- your To: header should be: To: sip:1011 at 192.168.10.150:2051;line=9avrmhew;intercom=true Mind you, I didn't get the phone to respond to the intercom=true until I added it on the request line as well, so the INVITE line of your request would be: INVITE sip:1011 at 192.168.10.150:2051;line=9avrmhew;intercom=true SIP/2.0 I'm on a Snom 220 with SIP 3.56t, and I'm stuck on the very next step of the process -- answering the phone's challenge to the INVITE request. The wiki indicates that the Snom needs to challenge with realm=snom, but even if I add snom into our internal DNS so that I can set the registrar to snom (that being the only way I can see to change what the phone uses as realm), it still rejects the digest response. Anyone have this working with recent loads of SIP that can shed any light on this? I've email'd snom a few days ago but have yet to get a response. According to their web page, they have a new office as of April 1, and I got a response to a support request (on this very issue) today saying that they'd likely not be able to respond until people are settled into the new offices, so you'll likely have to be patient with them. -- Joshua P. Dady ___ 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
