Re: [Asterisk-Users] GXP-2000 MultiPurpose Keys
For future reference, I think the Grandstream config files can program any parameter that's included in the web interface. If you want to set something that isn't in the template, you can use view source on the web form to figure out the name of the option: the field names in the HTML are the same as the ones that go in the config file. p. On Sat, 2006-06-10 at 02:06 -0400, Daniel Salama wrote: Wow! Awesome. This template is much more complete than the one on GS's download page. Thanks, Daniel On Jun 9, 2006, at 10:26 AM, Gareth Blades wrote: Yes you can as long as you have at least the 1.0.2.13 firmware. I have attached the template. The multi-purpose key settings are at the end. On Fri, 2006-06-09 at 14:41, Daniel Salama wrote: Is it possible to program the multi-purpose keys on a GXP-2000 remotely via a TFTP configuration file? If so, what are the parameters to put in the configuration file? Thanks, Daniel ___ --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 gxp2000_config_1.0.2.13.txt ___ --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 ___ --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] new beta Grandstream firmware HT488_496_386
Experiences with the HT386s seem to be pretty variable: they work OK for some folks, and are virtually unusable for others. A couple of months back, I installed two HT386s side by side. One of them would lock up on an almost daily basis; I replaced it with an SPA2002, which has been much better. The other one, interestingly, has been much more reliable: it's crashed once or twice in the 8 or 10 weeks since it's been there, but no more than that. Both of them were running the same firmware, so the difference in behaviour must be due to either hardware revisions or some quirk of the use patterns that the two units were seeing. p. On Tue, 2006-03-07 at 08:40 -0500, Steve Jones wrote: I'm almost afraid to ask, but is the HT 386 known for having a lot of troubles? I just installed one at home about 2 weeks ago, and knock on wood, it's only locked up once, and this was when I was still in the process of tweaking the config to work optimally w/ [EMAIL PROTECTED] I can't say I'm entirely pleased with the slight echo and buzz I'm detecting, but so far it's at least worked.. This isn't the consensus though, huh?! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: sábado, 4 de Março de 2006 0:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 They promised me this for my POS 386 adapters that need to be rebooted every few days from lockups about 4 months ago. Gee I wonder if this will work. Probably not. On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote: http://grandstream.com/BETATEST/HT488_496_386/ ___ --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] Application Faxing using SIP
On Sat, 2006-02-18 at 22:04 -0800, Lee Howard wrote: Traditional faxing (not T.38) pretty much requires a lossless audio channel. Normally the best way to get this is with PSTN channels/lines through a Zap device. That said, VoIP channels can be configured such that they are also lossless. IAXmodem, for example, functions on the premise that an IAX2 channel passing over the loopback device will be lossless. I have also seen lossless SIP and IAX channels running over a WAN, but they were very specificially configured, and I wouldn't expect most connections with traditional VoIP providers to be anything near the kind of losslessness that is required for this to work well. I have a PRI terminated in a TE110XP card on my Asterisk box. Right now we are using a separate analog line for faxing, but (for a variety of reasons) I would like to switch to sending and receiving faxes over the PRI via Asterisk. What's the recommended way to do this? The three obvious options I can think of are: 1. Connect the fax machine to an ATA and have it speak SIP or IAX to * 2. Fit a TDM400P with FXS linecard into the * box and connect the fax machine to it directly. 3. Replace the TE110XP with a multispan E1 card, connect a channel bank to the second span, and plug the fax machine into that. Option 3 can be ruled out immediately for us due to cost. Option 2 is quite appealing, but I've previously been told that running multiple Zap cards in a single machine is not a good idea. Option 1 seems like the cheapest and easiest, but I have no idea how reliably faxing will work over an ATA. Thanks for any insight. p. ___ --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] HandyTone 488 ata?
On Sun, 2006-01-29 at 19:15 +, Phil Blundell wrote: Our HT386s are also a little bit prone to locking up and needing to be rebooted, but that seems to be a different problem: it occurs less often than on the HT488, and seems to be triggered by something to do with call transfers (which we never did with the 488). I've just bought an SPA-3000 to replace the HT488, though I haven't installed it yet. I'm hoping that I'll have a better experience with this one. If that works out, I might toss the 386s in favour of SPA-2000s as well. In case anybody is interested, an update on this: I replaced the HT488 and one of the '386s with an SPA-3000 and an SPA-2002 respectively, and reliability does seem to be much improved. Neither of those units have crashed yet after a couple of weeks of use, whereas the '488 would lock up almost every day and the '386 about once a week on average. Early on, I had a bit of a problem with the SPA-3000 apparently not hanging up the FXO line properly at the end of a call; this seems to have gone away after some tweaking of the line settings, but I'm still keeping it under review. I also had a bit of a battle getting the dialplan on the SPAs working right, and I suspect there might still be a couple of things wrong there. The SPAs seem to be generally more configurable than the Handytones, as well. In particular, it looks like the ring cadences are configurable, which should be a welcome relief to those of my users who complained about the American ring cadence on the Handytone. Overall, I'm happier with the SPAs than the handytones, though neither of them are entirely perfect. Oh well. p. ___ --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] Application Faxing using SIP
On Sun, 2006-02-19 at 07:36 -0800, Lee Howard wrote: This will work provided that you can create a near-lossless communication path between the ATA and the PSTN gateway (which is the Asterisk box, I assume). One way of creating that, I would expect, would be to add another ethernet card to your Asterisk server and then run a crossover cable between that interface and the ethernet interface of the ATA. You'll also need to configure the ATA to not do lots of things typically done by ATAs, like echo cancellation. That's a good idea. I hadn't thought of using a crossover cable and a dedicated card like that. (Though, that said, I suspect that the datapath through our regular network switches is probably close enough to lossless for this purpose as well.) Any recommendations as to which ATAs are suitable for this purpose? I don't remember seeing a way to disable echo cancellation on either the Grandstream or the Sipura ones that I have here. p. ___ --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] attended call transfer
Probably because this isn't the way that a lot of other PBX systems work. It's not always easy to educate users about the difference between a blind and attended transfer when the systems that they've used in the past don't make this distinction. Disconnecting the outside caller certainly doesn't sound like a desirable response if you hang up the transferring extension while the transfer destination is ringing. I would kind of expect it to either degenerate into a blind transfer or return the call to the transferring extension; I can't, offhand, think of any likely situation in which I'd want to summarily hang up on the caller at that point. p. On Sun, 2006-02-12 at 10:57 +0200, Rob Lith wrote: Why don't you think it is correct behaviour? The purpose of attended transfer is that you consult with the party before transferring with hooking, otherwise it would be a blind transfer for which there is a blind transfer option. Rob On 2/10/06, Moises Silva [EMAIL PROTECTED] wrote: this is a Normal behaviour, nevertheless i dont think is a correct behaviour. Several weeks ago other user asked the same, i suggested him to open a feature request on bugs.digium.com, check for that regards On 2/9/06, Thomas Artner [EMAIL PROTECTED] wrote: Hi! I am new with asterisk and I have my first problem with the attended call transfer feature. When a call comes in, i take the call and i would like to transfer it. So I press the * button (mapped for the attended transfer in features.conf) and the number for the receiving extension. The receiving extension rings and the call can be taken there. So far so good. Now to my problem: If I hook on the handset BEFORE the receiving extension take the call, the caller from outside will be disconnected and the receiving extension stops ringing. Shouldn't the receiving extension keep on ringing until the call is taken? Independent of hooking on the handset or not! (as it is with the blind transfer feature) The incoming line and all of the extensions are POTS, connected on a tdm400p card. I use asterisk 1.2.4 and zaptel 1.2.3 Hope someone could help me. Thx, Tom ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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 ___ --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] ddi???
On Sat, 2006-02-04 at 23:33 +, Chris Bagnall wrote: You need to get BT to agree and allocate or port the numbers. You need to agree how many digits BT will pass on to you (probably 1925838395 but possibly just the last 2) I don't know the number of digits that BT pass through on a PRI, but on a set of BRIs with a range of DDIs, they're passing the last 6 digits (so given the OP's range, you'd want to match on 838381 etc.) BT generally like to pass 6 digits in and out of their network. You can request to have fewer digits sent; I'm not sure if they will let you have more. p. ___ --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] changing displayed call info on snom 360
Thanks for the suggestion. After exchanging some emails with snom support, it turns out that the 360 firmware doesn't quite support the facility I was looking for. My actual problem was that the To: field in the body of the INFO needs to match the identity of the line in question. If I make that so, I can indeed use the From: field in the INFO body to change the calling name and number that the phone reports. But this doesn't really help me much, since I could have changed that anyway by just tweaking Asterisk's caller ID variables before sending the original INVITE. I was hoping that the INFO message would cause the 360 to display the name and number for both the calling and the (originally) called party. p. On Mon, 2006-01-30 at 11:02 +0100, Christian Stredicke wrote: That INFO must be inside the extsting dialog, maybe that was the problem. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Blundell Sent: Monday, January 30, 2006 10:16 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] changing displayed call info on snom 360 Several of my SIP users are in the habit of diverting all their calls to an assistant when they're out of the office. When these calls ring on the assistant's phone, she wants to be able to tell which number they've been forwarded from so that she can say Joe Blow's phone or whatever when she picks up the call. The assistant's phone is a snom 360, which normally just displays the number of the calling party while it's ringing. Snom's FAQ page at http://www.snom.com/wiki/index.php/FAQs suggests that I can send a SIP INFO message to the phone to change the displayed call information. I did a few experiments with a hacked chan_sip.c, but wasn't able to produce any visible effect on the phone. Does anybody have any experience making this snom feature work with Asterisk, or know of any other way to influence the information that's displayed on the phone? Thanks Phil ___ --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 ___ --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] international caller id on UK (BT) PRI
When a call arrives on our PRI from a UK domestic number, the presented caller ID looks something like 1223123456. In my dialplan, I stick 90 on the front in order to turn this into a valid number for outward dialling, and everything works fine. However, when a call comes in from an international number, I need to add an extra zero -- that is, 491234123456 needs to have 900 added on the front to make it valid. Is there some Asterisk variable I can inspect to find out whether the presented CLI is using a national or international number plan? thanks p. ___ --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] Grandstream Budgetone mass deployment?
On Mon, 2006-01-30 at 11:42 +0200, Dmitry Ivanov wrote: I am considering mass deployment of Budgetones 102. According to their website, remote provisioning (configuration via TFTP) is possible. Anyone has experience with this? Is this really working? It does work, yes, though I think you need to configure the TFTP server address manually on each phone. Personally I'd be a bit wary of mass Budgetone deployment for other reasons, but the remote configuration stuff shouldn't be a problem. Grandstream use basically the same configuration file system for the Budgetones as they do on the Handytones and the GXP-2000. Obviously you need some way to make the files in the first place: when we deployed our GXP-2000s I ended up writing a little Python script to create the Grandstream config files (and the associated Asterisk config entries) from input data in a Gnumeric spreadsheet. p. ___ --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] Grandstream Budgetone mass deployment?
On Mon, 2006-01-30 at 15:14 +0200, Dmitry Ivanov wrote: On Monday 30 January 2006 13:03, Phil Blundell wrote: Personally I'd be a bit wary of mass Budgetone deployment for other reasons, but the remote configuration stuff shouldn't be a problem. What reasons do you mean? Just that, from my limited experience of Budgetones, they seem to be generally a bit buggy. But if they work OK in your environment, there's probably no reason not to use them. p. ___ --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] Grandstream Budgetone mass deployment?
On Mon, 2006-01-30 at 11:51 -0800, [EMAIL PROTECTED] wrote: does your python script generate the binary format grandstream files or do you still need to use their closed-source tool? Right now I'm still using their Java thing, but it's slow enough that one of these days I guess I'll crack and reimplement that stuff directly in python. I think the algorithm is described on the voip-info.org wiki someplace. p. ___ --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] HandyTone 488 ata?
On Mon, 2006-01-30 at 23:46 +, Chris Bagnall wrote: Has anyone ever gotten that working? I've tried it on every Granstream device I've had (budgetone, HT486, GXP-2000) and it's far from reliable on any of them. Seems that when dialling an external number, the phone accepts the first 3 digits (on a 3-digit extension dialplan), then refuses to accept any more digits. Yeah; it's working great on my GXP-2000s (with the 1.0.1.13 firmware). I don't remember what exactly was going wrong with it on the Budgetone and Handytone. I seem to remember there was some bad interaction with SIP authentication, and that setting Asterisk to not require the phone to authenticate itself improved matters a bit. Maybe older versions of the GXP-2000 software were broken in the same way; what version do you have? p. ___ --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] changing displayed call info on snom 360
Several of my SIP users are in the habit of diverting all their calls to an assistant when they're out of the office. When these calls ring on the assistant's phone, she wants to be able to tell which number they've been forwarded from so that she can say Joe Blow's phone or whatever when she picks up the call. The assistant's phone is a snom 360, which normally just displays the number of the calling party while it's ringing. Snom's FAQ page at http://www.snom.com/wiki/index.php/FAQs suggests that I can send a SIP INFO message to the phone to change the displayed call information. I did a few experiments with a hacked chan_sip.c, but wasn't able to produce any visible effect on the phone. Does anybody have any experience making this snom feature work with Asterisk, or know of any other way to influence the information that's displayed on the phone? Thanks Phil ___ --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] HandyTone 488 ata?
On Sun, 2006-01-29 at 12:36 -0600, Rich Adamson wrote: Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Anyone with good experiences? I played with FXO on the HT488 a bit, but didn't have a whole lot of luck. We had a bit of a problem with echo, but more seriously the thing kept getting itself into a variety of wedged states: sometimes it would lock up altogether (usually with its button lit up), and sometimes it would refuse to auto-answer calls coming in on the FXO interface. These latter problems have been severe enough that I didn't bother trying to diagnose the echo thing. Plus, even when set to auto answer after 1 ring, it often seemed to wait for three or four rings before picking up. Our HT386s are also a little bit prone to locking up and needing to be rebooted, but that seems to be a different problem: it occurs less often than on the HT488, and seems to be triggered by something to do with call transfers (which we never did with the 488). I've just bought an SPA-3000 to replace the HT488, though I haven't installed it yet. I'm hoping that I'll have a better experience with this one. If that works out, I might toss the 386s in favour of SPA-2000s as well. p. ___ --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] HandyTone 488 ata?
On Sun, 2006-01-29 at 13:24 -0600, Rich Adamson wrote: Seems the spa3k functions pretty well (had a few since they first came out), but the echo can on long analog loops leaves some to be desired as well. Short loops seem to work just fine. Thanks for the information. Sounds encouraging. The only significant feature that the SPAs seems to be missing compared to the HTs is the Early Dial thing (where it sends each digit to Asterisk until it gets something other than a 484 response back). Without that, my users need to either wait for a timeout or dial #, neither of which is terribly appealing. However, I haven't ever quite succeeded in making Early Dial work reliably on the HTs in any case (although it seems to be fine on our GXP-2000s), so I guess it's not as much of a loss as all that. And, well, I guess I can live in hope that Sipura will implement this one day. p. ___ --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] wildcard matching in dialplan
On Sun, 2006-01-22 at 18:18 +0100, Wilson Pickett wrote: You could also use a trick like *21* going to a new context and waiting for digits (with a slighly longer timeout) and have it trigger on the longest possible number. perhaps if local extension were of the form 2nnn or 2nn and you want to use both local and normal local POTS numbers you can use two or more extensions: _*21*2xx* _*21*2xxx* _*21*nxxn* etc. Use the include= trick to prioritize the last three properly. Thanks for the suggestions. Unfortunately your second option wouldn't have worked for me because my users want to forward calls to all manner of external numbers, including international ones with unpredictable formats. I thought about doing the first thing you mentioned, but I wasn't sure whether this would work if someone programmed a redirect into a speed-dial key and Asterisk got hit with the whole number at once. Anyway, I ended up hacking the pattern matching code in pbx.c to support the kind of patterns that I needed. It's a bit gruesome, but seems to be working well enough. p. ___ --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] wildcard matching in dialplan
I'm trying to write some dialplan patterns to allow my users to control
call forwarding from their handsets. Right now, I have this in
extensions.conf:
[forwarding]
exten = _*21*X.X*,1,Macro(set-cfim,${CALLERIDNUM},${EXTEN:4})
I was hoping that this would match any string of the form *21*nnn*,
where nnn is 3 or more digits. But Asterisk seems to ignore the final
X*, and allows it to match strings of any length that start *21*nn.
This is a problem because many of my users are using GXP-2000s with
Early Dial enabled: I need Asterisk to go on rejecting the number with
484 address incomplete until it sees the final * digit.
Can anybody give me a clue how to accomplish this?
thanks
p.
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