Re: [asterisk-users] Dealing with 2 SIP providers
On Fri, May 11, 2007 at 11:06:35AM -0400, Mike wrote: Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available What would be really cool, but require special code in the chan_sip dialer, would be automatic support of multiple providers in a similar fashion to the way Asterisk can ring two channels and only talk to the first to answer. You can't just do this with outgoing providers, because if you try to ring two at once, you may very well have the second one go to a voicemail and thus answer right away (because the first is ringing) and you would treat that as the success. What I have in mind is something like this: a) Invite to main provider b) Await some intermediate response, such as a RINGING code or some early media c) If you don't get that after a short timeout (more like 5 seconds) then INVITE the second provider d) Upon the receipt of a ringing or early media code from either, CANCEL the other. Now you would have to get your timings right because there could still be risk of doing something bad, such as a 2nd call going to voice mail or residual ringing making a call waiting on the recipient. (I don't know what typical 5ess do with a 2nd call that comes in while still ringing, anybody known?) Anyway, this could be a good course when a provider has known unreliability. Long timeouts and restarts are very annoying to 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] Re: Refund from SellVoip?
On Sat, Mar 24, 2007 at 12:13:25PM -0700, Martin Joseph wrote: On 2007-03-23 14:37:18 -0700, Tom Lynn [EMAIL PROTECTED] said: Now I know where they've been spending my remaining balance... I still use Sellvoip as my primary terminator, and have found the call quality to be superior to any other ITSP from my location (Seattle). I agree completely that there is no support from this company, which is a major issue if you are trying to support other customers. Still, I remain a happy customer of sellvoip, with Teliax and Nufone configured as backups... I wouldn't expect a refund for cancellation of prepaid phone usage, does the original agreement you have with then suggest that they owe you a refund? Sure, if you had a bunch of DIDs with them, and you sent in requests to cancel them and they never acted on them and kept billing until your accounts ran out of money. Or if due to a bug, they started billing another customer's calls to you, quickly depleting your account and giving you access to that other customer's private CDR data, and they never answered any tickets, calls or emails about the matter for days. Yes, I could see wanting a refund. The good call quality and no customer service routine seems good, until something goes wrong and you have no idea when they will fix it. You can use backup providers for termination, but not origination. ___ --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] Nomination for Coolest App in 2007
On Wed, Mar 14, 2007 at 09:37:45AM -0500, Steve Totaro wrote: Another interesting (from an American's perspective anyways) is that inbound calls on cell phones are free. Even if you buy a SIM with a little pre-paid time and use up the time, you can still receive inbound calls for free for a couple months. Inbound calls on cell phones outside North America are alas, not free, though people pretend they are free. They are caller pays for airtime. The only free incoming call systems I have seen are some mobile to mobile free call plans, and a small number of North American mobile plans that, for a flat monthly or daily fee, offer free incoming. The caller-pays system found outside North America is, in my view -- though I know some differ -- one of the last, great curses of old world telephony on our new environment. With my VoIP terminators, I can call most of the world's landline's for a price so low I think of it as free, with one exception -- the damn caller-pays cell phones which cost over an order of mangitude more because the fact that the payer doesn't negotiate the price removes the competition that would normally drive the price down. (And has driven it down in the receiver-pays countries.) However, for people in those countries, the bluetooth module does seem like a good idea. Obviously in places with no landlines, but also in places with these bizarre prices, so that if you call one mobile from another mobile, it's cheap, but if you call from a SIP terminator, it's 25 cents/minute. ___ --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
Bluetooth Re: [asterisk-users] Nomination for Coolest App in 2007
Another idea that has just come to me regarding bluetooth and a PBX is like this. Many people would like to use headsets with their IP phones. Some support wired headsets, but bluetooth headsets can be a good choice for a headset -- no wires, many people often have one, and there is a rich competitive market that makes them cheaper than many of the headset products available.There are a few hardphones that will take a bluetooth headset -- this seems to me like an obvious idea that's not very expensive to implement -- and I have a plantronics bluetooth base station for use with any phone that works on my Cisco 7960 and other phones with a headset jack. But it's expensive, and in the latter case, goes digital-analog-digital. So with a bluetooth channel that can talk to a bluetooth headset, you could have Asterisk itself give you your bluetooth headset on your desk phone.To do this, you would: a) Send calls to your desk phone to both the phone and headset. You can answer on either. See caller id on phone. b) Tranfer calls from desk phone to bluetooth headset. (Unfortunately requires the cumbersome transfer function of many phones.) c) Better still, have it so if the bluetooth headset opens a connection, and the paired desk phone is in a call or has a call on hold, auto-grab that call and put it on the headset. d) If the BT headset hangs up, instead of a normal hangup, consider that a transfer back to the desk phone, which will ring, and can then take over the call for any phone functions (real transfers, etc.) If you really meant to hang up, it does mean you have to answer and immediatly hang up this final call.The user could program if she wants this, and from which modes she wants it. Other than some minor inconveniences of (d), you get something much like the ability to have a bluetooth headset as a handsfree headset for any phone on the system, even analog phones. Unfortunately, you can only have a limited number of BT headsets operating at once from any bluetooth dongle (typically 8, and at that point you also get interference issues.) However, dongles are only $10, so you can have several on one server. If people will be far from the server, you have to do this functionality from remote machines, which might be best done with an IP phone softphone module. In that case, you can have more UI, include a choice on termination. The general idea of a secondary phone which, if it connects, automatically grabs any call on the main phone is handy. For real phones with dials, you can have this be a magic extension to dial. Bluetooth headsets can't dial and can have it simply happen on connection. ___ --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] Refund from SellVoip?
On Fri, Mar 16, 2007 at 11:32:31AM -0700, Tom Lynn wrote: Has anyone been successful in getting a refund from SellVoip when you've cancelled service? You were able to cancel service with Sellvoip? That's impressive, that implies they actually responded to a request you made to cancel service. It was rare I could get them to respond to any request or ticket before I gave up. Their service (as in call quality) was pretty good. It's a shame they could not, while I was using them, afford to provide customer service. It's one of the few companies I ever wrote to to say, Could you please raise your prices, so that you can afford to hire a customer service rep? ___ --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] Refund from SellVoip?
On Fri, Mar 16, 2007 at 04:16:21PM -0700, Tom Lynn wrote: At this point, I'm simply contacting the State of Washington Attorney General's office. They're ignoring my e-mails and I'm done monkeying around. It makes no sense. The put together a good system on the tech end, Asterisk based, decent call quality and faster call completions than any of the other folks I have been trying, at good prices. And then dropped it all on the floor, not responding to calls, emails or tickets often for weeks and months, if at all. Their interface needed work but that I can tolerate. Not being able to reach somebody for an urgent problem makes no sense. Does anybody know Jed Stafford? As far as I can tell this ended up being a one-man or two-man operation. It's just sad. ___ --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] Nomination for Coolest App in 2007
On Tue, Mar 06, 2007 at 11:14:15PM -0500, Steve Totaro wrote: Mine goes to chan_bluetooth. Somewhat of a pain getting it going but I am totally floored with how cool it is! Thanks, Steve Totaro ___ --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 My, that is a cool app. I look forward to running it when it's a bit more stable. While the outgoing call ability seems of limited use since cell call quality is not that exciting to even unlimited night and weekend minuets are probably not too attractive compared to 1 cent/minute SIP terminations, there are a number of interesting possibilities: a) If your target has an unlimited calls to other customers/family/etc. plan, you would want to call them this way to save minutes. b) Handy on some carriers for checking cell voice mail. (I have found that with many US carriers, however, you can call your cell phone with CID set to your cell number, and it goes directly to voice mail Make sure you have a password!) c) Incoming calls, obviously handy. d) During daytime, program to receive incoming calls and say, I am at my desk. Please call me at xxx- or press 1 to have me call you back at CID so you get better quality and don't bill cell minutes. In the evening, assuming unlimited weekends, you might forward directly. Can it send and receive SMS via bluetooth too? I also like a lot the talk of coming softphones with bluetooth headset support. This would allow you to use your bluetooth headset as an extension on your Asterisk pbx. I happen to have a bluetooth headset that plugs into my hard phone -- I wish more hardphones supported them natively -- and that's handy. This could be just as good. To really get it right you would want some speech recognition so you could place calls from the bluetooth headset by saying names and digits, as many cell phones can already do. Of course, a linux softphone could reside right on the asterisk box. You could multi-dial your bluetooth headset and your hard phones and answer where you like. ___ --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] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
Do I get 1.4.1 by just doing an svn up in a checkout of the 1.4 branch, or do I have to switch to a new tag or branch for what I have checked out? I did an svn up and there are new files, but nothing in the change files about it being 1.4.1.Many packages with various minor versions tend to have the master branch (like 1.4) mean The latest stable version of 1.4, be it 1.4.0 or 1.4.whatever, while if you check out 1.4.1 that means you stay at 1.4.1 even if there is a 1.4.8. What's the procedure here? ___ --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] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
On Sun, Mar 04, 2007 at 08:50:48AM -0600, Kevin P. Fleming wrote: Brad Templeton wrote: I did an svn up and there are new files, but nothing in the change files about it being 1.4.1.Many packages with various minor versions tend to have the master branch (like 1.4) mean The latest stable version of 1.4, be it 1.4.0 or 1.4.whatever, while if you check out 1.4.1 that means you stay at 1.4.1 even if there is a 1.4.8. You will never see ChangeLog files in the branches in our Subversion repository, because we only create them in the tags as we make releases. Since you didn't give us the output of 'svn info', we don't know what you have already checked out... but if you checked out http://svn.digium.com/svn/asterisk/branches/1.4, then you have everything that is in Asterisk 1.4.1 plus whatever changes have been committed to the 1.4 branch since the release was made. Thanks, Kevin. Yes, I have http://svn.digium.com/svn/asterisk/branches/1.4. I was running /trunk before but it wasn't stable enough to be a production system (no surprise.) I am presuming that the above is intended to be stable in this fashion. In many packages there is some file (usually the change log) which always tells you what version of the program you have in your hands, in terms of the program's current version number -- of course you can see the svn revision numbers and dates but they don't trivially translate. ___ --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] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
On Sun, Mar 04, 2007 at 02:34:21PM -0600, Kevin P. Fleming wrote: Brad Templeton wrote: In many packages there is some file (usually the change log) which always tells you what version of the program you have in your hands, in terms of the program's current version number -- of course you can see the svn revision numbers and dates but they don't trivially translate. I would be surprised to see such a file in a direct checkout from the project's SCM system. Even if that file existed, it would exist for a very short time as the moment a new commit occurred that branch would no longer 'be' 1.4.1, for example. Yup, typically it's a changelog and significant changes are noted along with version number bumps. I'm presuming the /branch/1.4 is the latest stable version of 1.4 with the latest patches.Since there is a 1.4.1 it means it is also 1.4.1 with the latest patches -- or so I presume. Having a file means people can look at see what they have, without having to ask here :-) Now that I know I can interpret what it means.I'm assuming that the latest /branch/1.4 is the one to run if you want a stable system with all known and tested patches and fixes but only modest new functionality -- or should one really be running /tags/1.4.1 and regularly updating your tag in order to get that? ___ --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] Sellvoip configuration....Please Help!!!!
On Fri, Feb 23, 2007 at 09:11:18AM +0100, [EMAIL PROTECTED] wrote: hi guy, i have a problem, i have an sellvoip account and i want configure asterisk for outbound calls. Alas, the best sellvoip configuration, I eventually had to conclude, was not to use sellvoip. They have good quality service, which makes this even more frustrating, but they are woefully understaffed, and can take months -- yes months, not hours, not days, not weeks -- to respond to support requests and tickets.They really are a good value when they work, but I had to abandon them, because problems can appear and you have no idea when they will be fixed. ___ --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] Open CallerID Database?
On Tue, Feb 20, 2007 at 12:08:15PM -0700, Natambu Obleton wrote: Why not make it like DNS and have each provider have their lookups deligated to a local server and then each ISP will run a caching server that will use a serial number system to get updates.. just like DNS. I know there are lot more DNS lookups then CNAM lookups per hour... isn't there? :) Hey, we could even build a system where DNS can be used to take any phone number and look up data about it, not just a name, but even a URI to redirect calls to for it, a source of presence info and more. What a great idea! Unfortunately, since phone numbers are believed to be owned by telcos and not by individuals, such a system would probably make the mistake of delegating control over the numbers to the telcos, who would feel no particular motive to help people bypass what they sell, and so I predict it will languish for a long time with no real deployment in the USA. :-) ___ --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] Open CallerID Database?
On Mon, Feb 19, 2007 at 09:02:56PM -0500, C F wrote: I doubt it's CNAM since it has old an outdated listings. On 2/19/07, Paul [EMAIL PROTECTED] wrote: Does google really have the true CNAM database? When I enter my number, I get a search result for my business listing at yellowpages.com Are you referring to something available in a google area other than the search engine? Well, at one time Google got a large telno to name database. I don't know if they have updated it. They can certainly afford to. There are other web sites that do reverse number lookups as well. Still, starting with their database seems a good choice. They might not like you scraping it at once but a thousand * boxes pulling records one call at a time is not something they are going to be bothered by. If this, combined with other info from other sources (including contributions from people who have CNAM) builds a workable database, you will eventually get the LECs contributing their data to it. People want their name to show up correctly. If millions start using a database, the LECs will want their data in it, especially if entry is free or near free for bulk entries. ___ --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] Open CallerID Database?
On Mon, Feb 19, 2007 at 03:01:50PM -0500, Paul wrote: I think terms of service for most CNAM providers prohibits sharing the data and limits the amount of time it can be cached for your own reuse. I don't know why they manage to get this level of control over the cnam database so that they can charge a penny per lookup as well as monthly fees. Does anybody know how this happens? Clearly some people buy the database at a good price. Google for example has it, and there are asterisk hacks to do google lookup (if you query a 10 digit phone number in google, you'll get not just name but address etc.) Perhaps they are just paying. One way to build a free database would be to simply have people share the results of all sorts of searches. People who pay for CNAM as end users, for example, have signed no contract to not share the data. So they could, if trusted, forward those records to be stored in the shared database. People who don't could take any number they get, and if it's not in the shared database already, do a google query, and if that gives a result, store that in the shared database. (Also store negative results with a timestamp so that you know that the google lookup provides no info.) http://www.google.com/search?q=nnpb=r Eventually you would get a pretty good database, perhaps one big enough that CLECs start wanting to update it directly? Now there may still need to be something to pay for all of this, but the fees could be much lower. Charge fees for the latest copy or real time query but just have the regular database out there for download and local lookup. Or perhaps just use the google api? ___ --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: NAT: RTP Path Optimization
On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote: PC When I set for Extern1/2 canreinvite=yes it works, but PC Intern-2-Extern doesn't work because Asteisk gives out the PC private IP-Adresses of Int1/2 Asterisk can't give out a public IP-address for Int1/2. Where would it get one from? Correct that it doesn't. But some kind sould could indeed code a variety of techniques to get it, such as: a) If the phone supports STUN or a static external IP mode, it often will include the known external IP in its SIP headers. Asterisk would ignore those headers and talk directly to the internal address (equivalent to what it does with NAT=yes, but for RTP too) when talking directly to the phone, but when connecting the phone to outside addresses, it would pass these addresses and SDPs. b) In many cases, such phones sit behind a NAT which can have hard coded port forwardings so that Asterisk could simply be told, in sip.conf, what the external IP and port of the phone are, and it could rewrite SDPs when needed. c) The phone could attempt to reach Asterisk through an external address, even though the phone is on the same natwork as the Asterisk. This would tell Asterisk the external IP of the phone, and it could rewrite SDPs. However, since many NATs don't support hairpin of audio, for those NATs asterisk must use internal SDPs to connect devices behind the same NAT, including itself if that is the case. Anyway, as noted, there are many answers to your question of how asterisk _could_ get the info from, but at present it does not. They are all kludges of course. Just about all NAT penetration involves some kludging. ___ --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: NAT: RTP Path Optimization
On Tue, Jan 30, 2007 at 12:00:17PM +0100, Patrick Cervicek wrote: Brad Templeton schrieb: On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote: PC When I set for Extern1/2 canreinvite=yes it works, but PC Intern-2-Extern doesn't work because Asteisk gives out the PC private IP-Adresses of Int1/2 Asterisk can't give out a public IP-address for Int1/2. Where would it get one from? Correct that it doesn't. But some kind sould could indeed code a variety of techniques to get it, such as: Again: My Problem is not Intern-to-Extern (NAT,Stun). My Problem is Extern-to-Extern, that the external phones are not talking RTP *directly* to each other. This is bad, when Asterisk is in Europe and the Phones are in Asia. ___ That you can usually make work with most phones today, as long as the phone has some NAT penetration in it.The most versatile approach is STUN -- see if you can configure STUN on your phones. Some phones that don't have STUN do support a hard-configured external address. However, they often require a numeric IP, which means that they only work if you have a static IP, and your RTP port number never changes on the outside. A few phones also support extracting information from rport fields etc. However, they tend to have STUN too. Then you must also keep a port in the NAT open, by one of the following methods. a) Manual hole in the NAT -- most NATs support this (port forwarding) and it is recommended. Set your phone to a fixed SIP Port and port forward that to your phone. Requires phone have a fixed internal IP usually. b) Keep alive transmitted by phone (will be on phone's config) c) qualify=yes in asterisk sip.conf sends keep alive from asterisk. Works, but if it ever fails your phone won't ring until it re-registers. d) Very short registration time, like 30 seconds to 1 minute. This will effectively do a keep alive from the phone. I like this least. If your phone does not understand NAT properly it will forward a useless SDP that is internal. This is what some Cisco phones do. For those, you are currently screwed. Some day Asterisk might support rewriting SDPs that contain unreachable addresses but it does not, at present, do this. ___ --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: NAT: RTP Path Optimization
On Tue, Jan 30, 2007 at 10:23:09PM +0100, Benny Amorsen wrote: PC == Patrick Cervicek [EMAIL PROTECTED] writes: PC But then all RTP Traffic of my internal phones will go over PC Asterisk. I want RTP to go Peer-to-Peer. == Intern-2-Intern PC and Extern-to-Extern should go P2P and Intern-2-Extern should PC go over Asterisk, see picture I understand what you want. I am telling you that you cannot get what you want, and the best compromise you can achieve. Either your internal-to-internal calls go direct, or your external-to-external do. Pick one. You can almost get it, if your NAT will hairpin, by having your internal phones all present their external addresses. Then all phones will appear external, and all can talk to one another (though there can still be port change problems on reinvite if you don't do explicit ports) -- but internal to internal (and all other internal involved calls) will go through your NAT box, not your Asterisk box. However, the NAT must hairpin audio, and a lot of them don't. Here is Cullen's latest chart of who hairpins: http://tools.ietf.org/wg/behave/draft-jennings-behave-test-results-03.txt Only a few do it, and only a few make his group A category at all. Of course, many of these boxes are not expensive so it may be worth switching, though many of us love the WRT54G and its clones because of the ability to use open source firmware. However, the bastard does not hairpin, I don't know if any of the new firmwares do. ___ --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] NAT solutions
On Sat, Jan 27, 2007 at 01:55:31PM -0500, Jerry wrote: On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote: Unless you are monitoring calls, want full CDR etc, then that's what you want anyway. CDR are not affected by how the audio flows. While technically true, I believe (it may have changed in 1.4) that if you allow reinvites, the signalling path follows the audio path, and you end up with reported calls lasting 3 seconds. So, if you want full (ie accurate as to the length of time) CDR, then I think asterisk has to remain in the call path. That would have been an odd bug. Signalling in SIP only moves when you do a REFER or similar. Reinvites can't change it. Having the signalling flow differently from the audio is a feature, not a bug, a very important one. With SIP INFO (or its planned successor) you can get the DTMF without having to get the audio, which is highly valuable. Right now Asterisk needs to stay in the audio stream to get DTMF, and that is one of the prime reasons it does. (The others are NAT, recording and meetme, the latter 2 of which should be a small minority of calls.) This is an important thing. Done properly, audio should almost never flow through the switching machine, or only flow for a portion of the call. The result can be orders of mangitude difference in bandwith requirements. ___ --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] NAT solutions
On Thu, Jan 25, 2007 at 10:19:06PM -0800, Yuan LIU wrote:
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2
If Asterisk1 can talk to Asterisk2 at trunk level, I'll be happy.
While I'm not sure of what tricks * plays at all levels, you
can certainly make this work if you have control of the NATs to
open ports, or if the asterisk servers know the address of their
partner and thus can keep the NAT open by sending keep-alives.
The way Jeff Pulver puts it, ICE has conquered the world :-) Would love
to learn more.
ICE is a methodology. You list every way you might be reached
(LAN, external addresses and addresses of outside relays) and the
other endpoint tries every way it can, ranked in order of quality,
and picks the best one. So if you're both on the same LAN it will
see that and use it. If you can't reach one another except through
a relay it identifies that and uses a relay. If, of course, you have
a willing relay.
(Skype solved that last problem :-)
Is this the concept of STUN? Does this also create latency (by adding an
additional leg in the route), packet loss, even jitter?
STUN is something else. Using a relay does indeed increase latency
(and thus echo) and may increase jitter and packet loss, though latency
is the big issue.
I should have used FWD as an example. One can't say it uses proprietary
clients. Does it stay away from voice path?
It provides a relay if one is needed. I don't know about today but
they started using jasomi boxes sold to deal with this question.
___
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Re: [asterisk-users] NAT solutions
On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote: For a remote phone, not on the same network as the Asterisk box (in which event the NAT worries are different) you definitely want to use the same protocol for the phone as for your term/orig provider. Otherwise you will be forced to hairpin your audio through your asterisk server, adding latency and wasting bandwidth and cpu for little reason. Unless you are monitoring calls, want full CDR etc, then that's what you want anyway. CDR are not affected by how the audio flows. Monitoring calls does require hairpin of the audio. Most people who are not call centers do not wish to monitor all calls or even more than few calls. (In fact in many states it is illegal unless you inform the other party, mostly limiting it to call center use.) If you had a call center * server in the USA hairpinning a call between India and the UK it would be really dumb, but even over shorter links it's dumb. I agree. Single SIP phones can usually be got to work behind a reasonable NAT router. And with some work could be made to work without special config with all but the rarest NATs. Hopefully in 1.6. For a single phone - you are quite right. For multiple phones, I'm not sure I agree - multiple SIP phones behind a NAT router is going to require some extensive config , or a SIP proxy in the router. Not really, other than the issue of NATS that won't hairpin between the phones. I have this situation, and our 2nd home I have 2 phones, on the * server at my main home. While I have linux computers at the 2nd home, it would be silly to put up a * server for the two phones if they can work through the NAT. It's not a big deal to tell the WRT54G I have to forward two ports to the 2 phones (well 3 if you include the wifi phone). There is no need at the 2nd house for intercom, so I would not put in a local server just for that. However, it does mean the remote location can't have SIP phones without things like STUN. Ah, but it isn't just asterisk you have to change - it is all the SIP implementations and all the routers :-) STUN is quite common in SIP phones, in fact the only major modern ones to not do it seem to be the Ciscos, though I have not tried the 8.0 firmware on them. ___ --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] NAT solutions
On Wed, Jan 24, 2007 at 11:09:21PM -0800, Yuan LIU wrote: From: Brad Templeton [EMAIL PROTECTED] On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote: In the meanwhile, use IAX, which understands about NAT pretty well. If you have multiple SIP phones on a LAN behind a NATing router, just put a small asterisk box on the LAN. It can manage your hairpin calls internally, save you bandwidth by trunking the IAX traffic to the central asterisk and avoid all the NAT hassle by using a single port (outgoing) and refreshing it often enough for the router to hold it open. Tim Panton www.mexuar.net www.westhawk.co.uk/ IAX is a fine protocol as far as it goes, however this answer is really not a workable one. There are only a few IAX phones, and they are not nearly as solid and full featured as the many SIP phones. There are some IAX termination and origination providers, but there are far more SIP providers. ... IAX is great but SIP is also a reality, and putting Asterisk into the just works category is a really important milestone. One I think that is intended to be improved a lot for 1.6. I have a really dumb question. It appears that Yahoo, MSN, AIM, you name them, they don't have a NAT problem, and some use SIP. I don't think they all stay in voice path, either. What takes? When you control both ends of the path, you can eliminate all NAT problems. Skype also deals almost perfectly with NAT (by using other nodes as relays if necessary) as does IAX. SIP was designed without much attention to NAT and it's had to be added on later and the different phones are all at different levels of implementation. Some time ago, actually, the SIP and SDP groups devised the ICE protocol for highly reliable NAT penetration, but it is still some distance from wide adoption, and I don't know when anybody will code up Asterisk adoption. Larger services like you describe often solve NAT by relaying traffic through their servers. They use a trick, that if they suspect an endpoint is behind NAT, they just ignore what they see in the SDP, and send all traffic back to the source port/host that the traffic comes from. For RTP, they wait for packets to arrive at the (external, routable) RTP port they provided, and send the traffic back there instead of the often unroutable address in the SDP. Asterisk, if you set nat=yes, will do step 1 (SIP traffic back to the source it came from, ignoring Contact header) but it does not yet do the same for the RTP. If it did, you would be unlikely to get NAT trouble on phone to Asterisk calls, or calls hairpinned through Asterisk. But you don't want to hairpin unless absolutely necessary. It costs bandwidth and adds latency. Latency no only makes calls annoying, it increases the chance of echo. ___ --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] NAT solutions
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote: In the meanwhile, use IAX, which understands about NAT pretty well. If you have multiple SIP phones on a LAN behind a NATing router, just put a small asterisk box on the LAN. It can manage your hairpin calls internally, save you bandwidth by trunking the IAX traffic to the central asterisk and avoid all the NAT hassle by using a single port (outgoing) and refreshing it often enough for the router to hold it open. Tim Panton www.mexuar.net www.westhawk.co.uk/ IAX is a fine protocol as far as it goes, however this answer is really not a workable one. There are only a few IAX phones, and they are not nearly as solid and full featured as the many SIP phones. There are some IAX termination and origination providers, but there are far more SIP providers. For a remote phone, not on the same network as the Asterisk box (in which event the NAT worries are different) you definitely want to use the same protocol for the phone as for your term/orig provider. Otherwise you will be forced to hairpin your audio through your asterisk server, adding latency and wasting bandwidth and cpu for little reason. In addition, many people just want to do things like give family or employees a phone they can take home, or take to a remote location and use on the PBX. They probably can't just put up an Asterisk server to make this happen, and nor should they want to. An additional server is not only more work and requires an always-on server computer, it's another thing that can go wrong. No thanks. Even if you can run Asterisk on a WRT54G, and thus don't have the $200/year power expense of a server, it's still not what you really want. IAX is great but SIP is also a reality, and putting Asterisk into the just works category is a really important milestone. One I think that is intended to be improved a lot for 1.6. ___ --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] OT: High Quality Wireless Headset for Cisco IP Phones and *
On Tue, Jan 23, 2007 at 02:01:43PM -0600, Tom wrote: Has anyone found a high quality wireless headset that works well with Cisco 7960 IP phones on an asterisk system? I tried the vxxi offering but the sound quality was pretty bad. Since these are pricey, I don't want to sample blindly. I've got one of the Plantronics bluetooth ones. It's OK, but frankly, with bluetooth hardware costing just a couple of bucks, you would think we should just see bluetooth becoming standard in every non-budget IP phone. People already have the headsets in many cases, and you can go digital all the way, and even rely on the headset's echo cancellation if you like. SNOM has a high end phone with this but otherwise it's been much slower to come than you would think. Alas, this doesn't really answer your question. ___ --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] NAT solutions
Some NAT problems you can solve, some you never will. Many modern phones have NAT support in them, via STUN, or a static external IP address. Most NATs also offer port forwarding, so you can open a hole for the SIP port in the NAT so all outside can reach it. (With port forwarding, you need a constant address for each SIP phone, so that means either static IP for the phone, or a DHCP server with the ability to always bind a device to the same address - the latter is preferable because you can move your phone to other networks more easily.) Many devices also feature NAT keep alive on the SIP port. That is a must if you can't open ports, but it sure generates a lot of annoying debug output when you turn on sip debug. Nothing beats a permanent NAT entry point though. Some devices, notably Ciscos, just don't support NAT as well. They don't have STUN, and while they may have a static external IP mapping, that's no good if your NAT itself has a dynamic address, as most home broadband NATs do. Asterisk, if you set nat=yes (or often even without that) will take incoming packets from a natted phone, and look at the incoming address, and send back to it regardless of what the phone says in its SIP headers. That's handy, but unfortunately it does not do the same thing for the SDP, so if the phone hands out an SDP with an unreachable address, Asterisk handles it badly. Some SIP gateways are smarter, and if they see an unreachable address in the SDP, ignore it and send to whatever address they get incoming RTP from. You'll have better luck connecting to such endpoints. Many termination providers do this, so you may find your phones can talk to the term provider, but not to other phones on the same * box. Many consumer nats will not hairpin audio. That means if you do all this work to rewrite the addresses in your SIP headers/SDP via STUN so you look like an externally routable device, and Asterisk hooks you up with another device behind your same NAT, you will get one way audio. I get this problem -- I have a * box at one location, with most of the phones (no problem for those) and some other phones at another location behind NAT. These phones can talk to the main location, but not to one another, due to the hairpin. What fun. A new method, called ICE, was drafted a while ago but is getting slow adoption. In ICE, devices are given a list of possible ways they could reach one another (directly, through nats, via RTP forwarders etc.) They try them all and pick the best. In the end it will always work through the RTP forwarders, but that costs bandwidth and latency. So far, however, support is limited. ___ --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] IAX vs SIP trunks between Asterisk boxes
On Sun, Jan 07, 2007 at 04:12:27PM +, Thomas Kenyon wrote: Brad Templeton wrote: For SIP phone calling * box, relay to other * box and out to SIP phone, you definitely want SIP all the way. Unless bandwidth between the * servers is a concern, then you're better off keeping the link between servers as IAX. (preferably trunked) The bandwidth of the audio stream dwarfs the bandwidth of signalling traffic by orders of mangitude. So in fact, I think this is exactly wrong. If bandwidth to or between the servers is a concern, that's where you most want to not be in the audio path. It is worth remembering in this sort of setup, often the phones at one site will not have a route to the phons on the other site, so the calls wont be re-invited off to the handsets anyway. If it's phone-on-NAT to phone-on-different-NAT, it typically will not work. That doesn't mean it can't work if bandwidth is important. I think the complete solution, not yet in Asterisk as I understand it is for Asterisk to be aware of both the internal and external addresses of a phone, and to connect internal phones with their internal addresses, but to connect internal phones to external endpoints through their external addresses. Ideally audio never flows through asterisk unless it's doing an IVR dialogue or otherwise explicitly wants it to. (In fact, ideally DTMF goes via SIP INFO or its successors so that Asterisk can listen to the DTMF without being in on the audio.) Flowing audio through your box costs not just bandwidth, it adds latency, and very slight extra risks of packet loss. Latency is the bane of voip calls, it also worsens echo. ___ --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] IAX vs SIP trunks between Asterisk boxes
On Fri, Jan 05, 2007 at 11:33:02AM +, Gordon Henderson wrote:
On Thu, 4 Jan 2007, Noah Miller wrote:
Hi Damon -
Can anyone comment on the overhead added when a SIP call comes into one
asterisk box, is routed to another with IAX instead of SIP, and is then
sent
to the UA from the second box with SIP?
DTMF passthrough issues?
I've got a client with sip phones on several different servers and
IAX links between the servers, so I guess that's pretty similar to
your setup. I've never bothered to check for overhead since it was
never an issue (all servers are P4 2.8 ghz or Xeon 2.8 ghz, 1GB ram,
with never more than 3-4 calls going through any one of the IAX
links). I can say that DTMF works fine in this setup.
I'm doing the same on 1GHz processors - CPU usage is virtually nil unless
there's transcoding going on (about 4% per GSM transcode)
ADSL bandwidth is more of a concern for me in these applications )-:
While it would be work to set up, you actually ideally want to
trunk with the same protocol being used by the external phones
or endpoints. When connecting a SIP to SIP call (presuming you
don't have annoying nat problems or have turned canreinvite off)
the audio should go directly from endpoint to endpoint and not
via asterisk.Ditto on IAX to IAX calls.
For SIP phone calling * box, relay to other * box and out to SIP
phone, you definitely want SIP all the way.
In some ways, an ideal solution would have two trunk connections
between the boxes (really just two config entries in iax.conf and
sip.conf) and go between the boxes with whatever protocol the
calling channel is using. You could write dialplan scripts to
pull out the channel and choose the right * to * protocol (as
opposed to inter-asterisk protocol which has another meaning.
:-)
It can also be worth having a termination provider that you
can talk to with both IAX and SIP, and sending them the call
with the same protocol the phone used.
Annoyingly, IAX and SIP channels use different interfaces
to provide the address, so you can't do
DIAL(${chantype}/[EMAIL PROTECTED])
A cute patch would be to support that with a consistent syntax over
channels.
Note if you use various flags on Dial which require asterisk
to hear dtmf or do other audio, you are stuck hairpinning.
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Re: [asterisk-users] Re: Best inexpensive home office router for VoIP(QoS with maybe PoE)
On Fri, Jan 05, 2007 at 05:37:22PM -0500, Allen Casteran wrote: Mike wrote: You're quite right, I typed before thinking. Upload is the problem anyways, since it usually (in homes) uses much more limited bandwidth than downloading does. No answer to my question though: How do you people handle QoS without relying on the phones to do that? I'd like a box that can be purchased and installed easily (Linksys type of product) Mike, Unless your ISP specifically supports QOS on your internet connection there is NO QOS beyond your router. Only within your network will the QOS be effective. Once the packets go through your router all control is lost. :) This also means that you have little control over the priority of the traffic coming through the router's WAN port. The most you could do with QOS in this case is to limit outbound traffic from your PC if it would interfere with a voice call. The same is not true for the return (ie inbound) packets. True, but for many people the upstream path is the biggest, and sometimes the only bottleneck in their internet traffic, especially to a good termination provider that has not underprovisioned. So this is the one place QoS can make a difference. For downstream, it can be an issue. Though in theory a clever router can notice the amount of high-priority RTP traffic that is going through, and then cause incoming TCP traffic to back off to leave room for the RTP traffic. I don't know if the cheap boxes do this. The D-link DI-102 qos box literature seems to talk mostly about upstream so I don't think it does this. On the other hand, I tested my wrt54g with qos firmware on, and while downloading at full speed I detected no dropped packets in incoming voice, so perhaps it does that. ___ --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] DiD for less then $4
On Fri, Jan 05, 2007 at 02:29:03PM -0800, CM Rahman wrote: Anybody selling DID flat rate for less then $4 with sip or iax incoming? Let me know Thanks vbuzzer charges $2 for flat rate DIDs, not quite sure how they do it. However, I have had some clicks and pops of dropped packets with them I don't get with other DID providers, you should check how it does with you. And, worst of all, there is a bug in either their code or asterisk on reinvites. Asterisk hangs up the call after receiving the somewhat unusual reinvite OK that vbuzzer sends. However, if you don't need reinvites (ie. no IVR followed by xfer to phone with reinvite or other xfers) it could be good for you. The other question is, will you do more than 300 minutes (6 hours) incoming a month on average? If not, many people sell DIDs for $1/month and 1 cent/minute. ___ --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] 1.4 segfaulting when manager client is connected
I was just trying astman with the latest svn trunk from Dec 31. It
connects, but if I attempt to make a call, asterisk segfaults, but
in pthread_kill in /lib/tls/libpthread.so not in the asterisk code.
Is this something others have seen? This is with glibc-2.3.4-2
I just upgraded to 2.3.6 (the lastest for Fedora core 3) and it's
the same.
Not much of a traceback, it's happening here:
static struct eventqent *unref_event(struct eventqent *e)
{
struct eventqent *ret = AST_LIST_NEXT(e, eq_next);
if (ast_atomic_dec_and_test(e-usecount) ret)
pthread_kill(accept_thread_ptr, SIGURG);
return ret;
}
Should I file a bug on this? I would presume if it's as trivial
to duplicate as it is for me that others would have seen it.
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Re: [asterisk-users] caller id ring tones for Asterisk Phone
On Wed, Jan 03, 2007 at 11:15:19PM -0500, Jeronimo Romero wrote: I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Thanks in advance!!! ___ --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 Many phones can do this. Some have only a limited set of tones that don't vary much. Most phones can do the basics. Some let you have some uploaded wav file ringtones. A smaller number such as the SNOM phones and a few others can actually be given the URL of an audio file as the ringtone, and the phone will download it and play that. I haven't tried it, but it should be possible on the SNOM to: a) Have festival, cepstral or other TTS turn the caller id into an audio file (ideally cached) b) Put that audio file on a local web server c) Set the URL of the audio file as the ring tone. You usually set the ring tone with the SIP Alert-Info header, however various phones use different syntaxes. Do a search on voip-info for terms like ringtone and alert-info for instructions on how to set them. Of course, you can also do things like generate the audio and have your computer, or a nearby computer, play the sound so you get it reading the name or number. Or you could generate your own audio files for the people who call you regularly rather than that trick. ___ --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] SNOM 200 behind NAT and other xmas woes
I decided to give the whole family IP phones for christmas, all hooked into my asterisk server, so all the nephews can have their own lines. However, one of the phones I got was the SNOM 200. That's worked fine for me on my own network, but I'm having bad luck getting it to work behind NAT talking to Asterisk. It talks to my termination/origination provider, which seems to ruthlessly ignore SDPs and send audio to the address it gets audio from, which works pretty well behind NAT. I've tried all the various NAT settings on the SNOM 200 (with the last firmware rev they made) but reports are that's broken. The SDPs and Contact headers it sends out are always the natted address, even if I tell it to use STUN or static or UPNP etc. If the nat traversal is broken not much I can do on that end. Asterisk, on the other hand, should be handling this with nat=yes on the channel, but it's not. It handles it for the SIP packets, responding to those on the address the requests came from (ignoring the contact header) but it seems to accept the SDP, which contains and address Asterisk can't see. The docs say nat=yes will fix addresses in SDPs. I'm running svn trunk from a few days ago. Is there a way to get Asterisk to send audio to the address the incoming audio comes from, or to take the SDP and replace the IP address in it with the IP address the SIP came from? Otherwise while the phone will be able to make PSTN calls, it is unable to call Asterisk for voicemail and the rest. -- Some other issues: Anybody tried to use vbuzzer with Asterisk with an IVR on the DID? I find when I do this, after the IVR connext to an extension, the reinvite Asterisk sends to vbuzzer is responded to by a very simplified 200 state response which Asterisk seems to get upset at. Asterisk immediately hangs up the channel, but nothing in the debug logs (even level 9) seems to say why. I will use another DID for now. Also, I bought an SPA-2000 on ebay that claimed to be unlocked. It isn't. I will probably get my money back but that's not going to help at christmas. Anybody know a way to unlock these things at the hardware level (shorting pins etc.) The factory reset codes all demand a PW, which means it's locked. Web access disabled too. I have a Cisco ATA 186 but they only do NAT traversal with a static address, no STUN etc. I may end up setting up the SNOM and replacing it if no other solution shows up. ___ --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] SNOM 200 behind NAT and other xmas woes
On Sat, Dec 23, 2006 at 02:02:18AM -0800, Brad Templeton wrote: I've tried all the various NAT settings on the SNOM 200 (with the last firmware rev they made) but reports are that's broken. The SDPs and Contact headers it sends out are always the natted address, even if I tell it to use STUN or static or UPNP etc. If the nat traversal is broken not much I can do on that end. Well, I tried a bit more, including a numeric STUN server address and a reboot and that seemed to have helped. So now I'm down to the DTMF not working (I've tried both inband and rfc2833 and INFO settings but will try again.) And other issue. I have 5 line buttons set to 5 accounts. Incoming calls to these accounts light the right line button, and make the right ringtone. However, pressing the line buttons to make calls still results in all the calls coming from the first line's identity in the From: header. What's a good way to get the source line to identify on the SNOM 200? ___ --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] Dial own extension to get to voicemail.
On Wed, Dec 20, 2006 at 02:34:36PM -0500, Phil Finkler wrote:
I've gotten this Polycom 501 pretty much licked, but I need to know if
there's a way in a dialplan to say if someone dials their own extension
it goes straight to voicemail and asks them for their password. I
thought I saw an example of this on the web but I can't seem to find it.
Any advice appreciated!
You can do it, but it's more work than having an extension (the standard one
seems to be 86 now) that goes to:
VoicemailMain(s${CALLERID(num)[EMAIL PROTECTED]);
(But only in a context where the callerid can be trusted.)
To do what you want, you would need to have your extension processing
macro test if CALLERID(num) = ${EXTEN}, and then invoke the above
expression instead of dialing the extension.
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Re: [asterisk-users] NAT and Dial to two channels at once
On Mon, Dec 11, 2006 at 06:48:18PM -0500, Andrew Joakimsen wrote: You need to understand how NAT works, if you can chan2 and chan2 is behind a NAT and suddenly someone else is invited to chan2's IP address port 5060 chan2's router willl say WTF I dont have an estabished connection on port 5060 (to the client being reinvited to chan2) and it wont work. You need to have the media path go through asterisk in that case. Actually, it's more complex than that. If the NAT box has had a hole poked (in its config) for the RTP port (SIP port is only used by Asterisk) then any machine can send it RTP on that port. In addition, if the NAT is of the full cone type, any host can send to your port once you have sent a packet out that port. With Restricted cone and Port restricted cones, it also works as long as the Natted IP phone is sending packets out to the other host already. Which it should be if we have symmetric RTP. Symmetric NATs, which are rare, will change the port number when they start talking to a different host for RTP. This will screw up all but the cleverest implementations. (Though there are endpoints that notice if the RTP is coming from a port other than they were told, and start sending to that instead of the one in the SDP) What doesn't work is assymetric RTP with NAT. In this case we have the audio going through asterisk in one direction, and directly in the other direction. That will fail if the direct direction tries to go into a nat (it should work if it's only leaving a nat) Asterisk currently does assymetric RTP if it thinks it only has to listen to one end of the audio path. That's a good idea in general -- but not one that works through anything but a manually opened NAT. ___ --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] Cisco devices (without STUN) and dynamic NAT
Cisco devices (7912, ata-168, 7960 etc.) don't support STUN. However, they do let you define a static external NAT IP address, and parameters to send a keep-alive out through the NAT on a regular basis. However, I want to make these devices work in an environment where they are behind a NAT which has a dynamic IP which might change (though in fact it changes only rarely.) They're talking to an external Asterisk server which is of course not behind NAT. The docs say that nat=yes will cause Asterisk to ignore the IPs in the SIP headers and SDP, and replace them with the actual address the packets are coming from. I thought this meant that if I put any address in the Cisco's NATIP field, this would work because Asterisk would rewrite the SDP to the real address, which might have changed since the NATIP field was set. However, Asterisk is not doing this (though it's doing some other interesting things, now noticing that the NAT address in the Via header is local to it and talking directly to that) and it's trying to do native bridged channels to other devices, which isn't working with the wrong address in the SDP. I want native bridging of course, in fact it's a must if you have a phone on the east coast, an asterisk server on the west coast and a SIP terminator in the middle. No way you want to hairpin the audio, and with STUN supporting SIP phones this works fine. canreinvite is not involved here because there is no need for reinvite on a simple call. I'm using svn trunk, as of a few days ago. ___ --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] any possibility of Vonage Integration
On Wed, Dec 06, 2006 at 08:13:00AM -0500, Paul wrote: Also, I should have mentioned that many of these providers advertise business plans on their website. How can anyone honestly advertise phone, fax, email hosting, web hosting, etc. to the business community without 24/7 support? I like 24/7 support, but I would have to guess that most businesses would be mostly interested in support during working hours (which is more like 6am to 11pm for most companies.) Not that there aren't employees around at night sometimes, but I'm just talking about what's most important. I think with any company, Vonage included, it is good to have a redundant backup, for when their network or your network is down. Be it a PSTN line or a cell phone. Some companies offer PSTN failover on DIDs, which I think is a good idea. Works at least if your equipment, or their middle equipment is down but doesn't work if the PSTN failover equipment itself is down. Vonage does offer PSTN failover if your ATA is not responding. But having an FXO box talk to your Vonage ATA is just nuts. ___ --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] any possibility of Vonage Integration
On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) ___ --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] SIP firmware for Siemens Optipoint 410 Economy?
I have not seen anybody on the web to have found this so I thought I would check here. Anybody got this firmware? I've found firmware for the 400, but it doesn't seem to load in the 410. ___ --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] How to park calls on a specific extension
On Fri, Dec 01, 2006 at 04:55:51PM -0500, John Novack wrote: In most hybrid business systems one does NOT place a call on hold, but begins a transfer, either a specific function button or intercom button which automatically places the call on hold, gives a new dialtone and another extension is dialed. IF the called party answers, the transferrer can announce the call, and if the called party wants to accept the call, they simply hang up. Blind transfer is done the same Alas, it's not possible in these days to make it that simple. Today, almost all calls will be answered (by a voice mail) if not by the person. So you need an additional UI for attended transfer, which allows you to say No, I got the voice mail, disconnect the voice mail and bring me back to my call. I guess you needed that for endless ring in the old days too. You're right that there is no big difference between attended and unattended in the UI when it works, but when the attended call fails with voicemail or unlimited ring, or for that matter busy signal, you need a means to go back to the caller. That's one thing soft buttons are good for, you can create soft buttons for specialty functions like this. If you have line buttons on your phone, normally the original call is on one line button, and the 2nd call on another line button, so you just press the first line button to abandon the call attempt. On my Asterisk system, I have done another thing which is handy. My extension macro looks at the caller-id. Calls within the house do not ever go to voicemail. Calls from outside (including ones transferred) will go to voicemail after the timeout. So I never get voicemail but I do get endless ring. Many PBXs also offered a feature that if you blind transfer, and the call goes into endless ring that it transfers back to you after some timeout. Today, voicemail has largely eliminated that. ___ --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] How to park calls on a specific extension
On Fri, Dec 01, 2006 at 07:37:35PM -0700, Ken Williams wrote: I was able to set a program to speed dial the park extension. Then a user just hits TNFR followed by the line I've programmed to speed dial park. If you get the HOLD button to do this, I'd love to hear how :). Oh, that would require new code in Asterisk, a new commmand that is able to get all channels that are currently on hold, and connect to one if only one or give a menu and connect to one if more than one. Don't know if it would require any fancy changes to holding itself. ___ --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] How to park calls on a specific extension
On Thu, Nov 30, 2006 at 12:03:24AM -0600, Lacy Moore - Aspendora wrote: The question is what is the best interface? On our old system, we put the caller on hold, went to another phone, pressed pickup and then entered the extension where the call is on hold. I never liked that, especially if I was at an extension that wasn't mine. By the time, I got to where I needed to be, or someone called me and told me to pick that call, I would forget what extension. The same thing, I believe, will happen with the current park method. I don't know what would help with that, maybe better vitamins to prevent memory loss? :-) I don't know. Maybe a receptionist console that could tell who is on park, their phone number and caller id info along with who put them on park? If you integrate with the voice mail, so that you can pull a user's audio name for an extension, the pickup extension can say Do you want to pick up the call put on hold by 'Lacy Moore' or 'Joe Smith' or 'waiting room' or 'extension 242' Hopefully little need for memory. I'm wondering if maybe we are looking at having to have different ways of doing it. Being able to transfer the call to a line button, and being able to press that line button to pick up the call, and having the status shown, may be the better solution for small companies. Problem there is only some phones have line buttons, and when they have them they are scarce and there's many things you might like to do with them, and dedicating them to this would be low on my list. Dedicating one speed dial to a pickup call command that picks up the solo call or reads you the names/numbers of the calls on hold, or puts them on your screen if you have a screen -- that makes more sense, and it does well on every phone. Then if you want to have line buttons which read hints based on the number of calls held. I'm going to show my ignorance here. Since the phone displays the number we dialed,or the incoming caller information on the screen (we're talking those with displays), is there anyway to have it so that when the call is parked, it also shows the parking spot the caller is parked on? Kind of like hold does now? I know nothing about the SIP protocol, so I don't know if this is possible or not. Yes, some phones can receive text messages back from the server. Not all of them. But if you have a system where parking is just pressing hold, then all you need to know at worst is the name or extension of the phone you're on, and that's usualy already on the phone screen or even written on in pen! ___ --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] How to park calls on a specific extension
On Thu, Nov 30, 2006 at 02:50:21PM -0500, Tom Rymes wrote: for example: In your example above where they can't figure out how to transfer, why don't you edit features.conf and define the transfer key as # or something. Then, when they have a call for Bill across they way, they can do this: In this case don't they need to have a t in every Dial as well? And then there's the other direction. Sometimes (actually quite often) I like to transfer a call that I dialed, which requires the T but means you interfere with typing touch tones to IVRs that you call. No, almost all IP phones have a transfer button, the nice thing would be if somehow the UI for that could have been standardized, at least for the phones that don't have screens and soft buttons (which can extend the interface because they can show it to you). This is not generally Asterik's fault, of course. PBX interfaces are, as I said, notorious. Most users have no idea how to use most of the featurs on the PBXs they use, and a disturbing number don't even know how to transfer unless they do it frequently. That's where screen phones (or computer pop-ups) are a win. There are too many PBX interfaces that can't be improved. The nice thing about hold = park is that it is darn simple. The only thing it doesn't do is easily let you park on an extension whose number you don't know. The old parking lot can exist for that, I guess. Or just write the extension on the phone, or have an extension you can call that tells you your extension. ___ --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] How to park calls on a specific extension
On Wed, Nov 29, 2006 at 06:05:31PM -0500, Steve Sobol wrote: On Mon, 27 Nov 2006, Brad Templeton wrote: On Mon, Nov 27, 2006 at 11:20:27PM -0500, Andrew Joakimsen wrote: Can you explain how ValetParking and twenty minutes worth of dialplan creativitiy can't do the same EXACT thing you are describing? Sometimes the simplest answer is never the most obvious Yeah. With valet parking (or any parking) you have to explicitly park your call. With what I propose, or with SLA, or with many key systems or simple multiline phones, all you do is put the call on hold, and that makes it possible to grab it from elsewhere. Back on track... OK. I understand that press the hold button won't do what I want. The next best thing is ... instead of using the built-in call parking feature, where the call gets parked at a random extension, I need to be able to park calls from extension X at a specific other extension Y. Parking at a random extension, then picking it up, is fine if there's ever only one call on hold, but I expect that there will be times where I need to have more than one call on hold. We have four DID lines, all plugged into our Asterisk server, and we do a lot of telephone support. :) You could pull this off in a small system because the parking lot is big enough, I think with the valet parking add on. You can create a parking slot for each extension, and then grab them with a special extension. So if your extensions are say 30 through 39 you could use the valet add-on to always park extension 32 in slot 732, and then you could always pick it up that way. But I'm hoping that a better interface than lots can be devised with time. Lots are useful if you are a receptionist having to handle tons of calls, put more than one call on hold at once and pick them up in different unpredicted places. But otherwise their UI is not particularly good for ease of use. The best interface, I think, is either hold implicitly becomes pickupable after 5 seconds, or if you want a transfer, to have a parking lot you blind xfer to because you don't need to listen to the slot number, you just press a single button called park if possible. Let's face it, PBXs are notorious, even with new fancy screen phones, at being hard to use, and the UIs not remembered by their users, and also varying from phone to phone. There's much room for improvement in all PBX interfaces. Not just Asterisk. To my mind every PBX might do well just to have a single function button on the phone which causes a dialog box to pop up on the PC next to the phone, and do it all through that (though possibly by pushing buttons on the phone in most cases instead of the keyboard if you like.) A UI with clear buttons, grayed out buttons for what you can't do, help screens, warnings where appropriate, etc. SIP didn't go that way, it went towards the phone being full of features itself. ___ --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] SIP Port 5060
On Wed, Nov 29, 2006 at 04:49:38PM -0700, Joseph wrote: What I have is that each device is listening on different port ex. [pstn-5665] ; incoming/outgoing calls on FXO port type=friend ... port=5066 ; port on Pstn line ... [318] ; incoming/outgoing calls on FXS Sipura-2002 type=friend ... port=5069 ; port on FXS line ... etc. Though I'm not sure if bindport will work, try it! My understanding was that the port= field on a particular SIP channel defines the port used at the remote end, ie. The user's phone will be talking on port X of their IP address, it does not alter what SIP port Asterisk is listening on on the Asterisk box. That is what bindport does, and that's a global setting, I was not aware you could have multiple bindports but that is very useful if it works. The idea of having a different bindport per channel would be handy too as yet another means of identification. There is some merit in not using port 5060 for your bindport, though it comes at a cost. a) If you use another port, all clients must be configured with a slightly more complex URL for you including a port b) A few broken clients can't call you at all because they don't let you set the port (mythphone is one example.) but... a) You might get around carrier SIP blocking b) If there is ever a virus/DOS aimed at SIP devices or asterisk, you may well escape it. No guarantees becuase many mal-tools will scan for other ports but it's more work for them. c) If you want to run more than one instance it is the only way ___ --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] How to park calls on a specific extension
On Mon, Nov 27, 2006 at 01:46:58AM -0800, Steve Langstaff wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brad Templeton Sent: 25 November 2006 21:02 [snip] ...the UI I think most people want, which is, just put the call on hold, and go somewhere else and push a button to pick it up. That is not only an eaiser interface, it's actually a more powerful one, because it gives you the ability to put a call on hold to go away to check on something, and then decided after the fact that you want to pick it up from another extension. Indeed, you could create an interface if you wanted to so that you could pick up the call from any pstn phone (ie. cell phone) by dialing a magic number and entering a code, without having decided to explicitly park it first. I think that you are describing shared line appearances. I don't believe so, not as I understand them, but perhaps they are planned to be different from my conception in Asterisk. My understanding of a shared line is it mimics the traditional analog phones, a line is shared on many phones, calls ring on all phones, if you put on hold on one phone and pick up on another it works. What I describe is different. There are no shared lines, but if you put a call on hold on one phone on a non-shared line you can go to another -- any other in the pickup group, whether it is registered to have the shared line or not, and pick it up, as you can (in a more cumbersome way) with call parking. Or do I have it wrong, and with SLAs all lines will effectively be shared, and accessible from every phone? Both SLAs and what I describe allow you to do a very quick move from one extension to another by putting a call on hold and picking it up. However, what I describe is far more general.In some systems, shared line also includes multiple extensions ringing for a call on the line (already possible with Dial to multiple channels). It also means barging, in that I can pick up a shared line that is already in use and I join the conversation. The latter is done by shared lines and not by what I propose. Some PBXs have a barging system where you can say, Barge in on the call on extension xxx (this needs permissions typically.) This is also more general but not as intuitive as shared line if people all have phones with line appearances. In my home, for example, we really just have two people. There is no desire for a shared line. However, there is desire for a phone in the living room that rings whether her office line is being called or my office line. And there is desire for very simple call park, so I can pick up the call on the living room phone, and quickly put it on hold and go up to my office and pick it up quickly without the annoyance of current parking. I could do much of this with shared lines as I understand them but not in as nice a way. ___ --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] How to park calls on a specific extension
On Mon, Nov 27, 2006 at 04:05:34AM -0800, Steve Langstaff wrote: What I describe is different. There are no shared lines, but if you put a call on hold on one phone on a non-shared line you can go to another -- any other in the pickup group, whether it is registered to have the shared line or not, and pick it up, as you can (in a more cumbersome way) with call parking. I'm a bit unclear on where you say 'push a button to pick it up' - does this mean that there can be only one held call 'shared' between the extensions, or is there some logic somewhere that 'knows' which held call should be picked up when you press the button? In my view of the SOHO environment, you would put a call on hold. That's one button on most phones. At the target phone, a speed-dial button would be configured to call the pick up held call extension. If there is only one call on hold in the pickup group of that extension, it would simply connect that call. Let's say the pickup extension is 600. If there were more than one held call (quite rare in a home PBX) it would instead say, There are calls held for extensions 123 and 456. Please enter the extension you wish (followed by pound sign if there are ambiguous patterns like extension 22 and 222 but hopefully nobody does that! Otherwise you have to wait a few seconds after 22 is pressed.) In addition, you could also define so that extensions of the form 600xxx are a hard pickup of a call held by extension xxx. Thus if you want to be more reliable, or have a speed dial aimed at picking up only a very specific extension with no chance of a menu, you could do that. I would implement this with a PickupHeld command, which can take an extension argument, or no extension (meaning pickup any or give menu), or possibly a pickup group argument so dialplans could allow pickup from other pickup groups if you want to allow that for security reasons. Anyway, for the user, the UI is very, very simple, especially in a SOHO where mostly we're talking one call on hold at a time. For security reasons, as noted, you would not be able to pick up a call that was just put on hold in the last few seconds. And an extension could define if it wanted that calls it puts on hold are not available for remote pickup to avoid any risk of accidental pickup. Even then, they might not want to use the parking lot system (which has no real security) but just have to do an explicit transfer (like the xfer to 700, but no need to wait for a number to remember) For a large PBX serving several offices, you might want to expand the pickup group concept to have a master grouping, or allow pickup group numbers to be versioned. Ie. If I say pickupgroup=2.1 then 2 would define my company, and the 1 would be the traditional pickup group function within the company. Just a thought on a good UI. Obviously what's really more valauable is code. ___ --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] How to park calls on a specific extension
On Mon, Nov 27, 2006 at 11:20:27PM -0500, Andrew Joakimsen wrote: Can you explain how ValetParking and twenty minutes worth of dialplan creativitiy can't do the same EXACT thing you are describing? Sometimes the simplest answer is never the most obvious Yeah. With valet parking (or any parking) you have to explicitly park your call. With what I propose, or with SLA, or with many key systems or simple multiline phones, all you do is put the call on hold, and that makes it possible to grab it from elsewhere. Frankly, call park by transfer requires that the user be comfortable with transfer. You may be aware that quite often they aren't, and in fact, transfer doesn't always even work on some phones or takes work to get going. On some phones it's #700. On some it's hit xfer, dial 700, hit xfer On some it's put on hold, then dial, then hit xfer. Etc. etc. almost all phones however, that can put on hold. do so with a single well marked button, same UI everywhere. ___ --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] How to park calls on a specific extension
On Sat, Nov 25, 2006 at 07:17:47AM -0500, marvin horst wrote:
The valet system gets us partway from what I read, but it still uses the
arbitrary number slots. It still requires the user know to transfer a
call to the valet.
no you can park to a specific number (lotname)
exten =
_6XX,1,ValetParkCall(auto|8${EXTEN:1:2}|180|${CALLEDEXTEN}|1|internal)
; Valet unpark from an extension
exten = _5XX,1,Playback(beep)
exten = _5XX,n,ValetUnParkCall(filo|8${EXTEN:1:2})
Definitely better, and I will install this add-on, but still not at the
UI I think most people want, which is, just put the call on hold, and
go somewhere else and push a button to pick it up.
That is not only an eaiser interface, it's actually a more powerful
one, because it gives you the ability to put a call on hold to go
away to check on something, and then decided after the fact that you
want to pick it up from another extension. Indeed, you could create
an interface if you wanted to so that you could pick up the call from
any pstn phone (ie. cell phone) by dialing a magic number and entering
a code, without having decided to explicitly park it first.
The other reason it's superior is that call transfer differs on various
phones, and sometimes transfer to the parking lot doesn't work right,
it's one more thing to go wrong. However, almost all phones have the
same interface for putting a call on hold (a hold button) and it is
more likely to work.
The only downside to the implicit park UI is that somebody else can grab a call
you put on hold that you didn't intend to park. That's not an issue
in a house or small office, though. And it's low risk. For example,
you can wait 5 seconds to put a call into implicit park so that if you
are putting them on hold to do attended transfer, this can be spotted
so that there is no implicit park. Implicit park would require you put
the call on hold and do nothing do it for some amount of time if you
like.
___
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Re: [asterisk-users] How to park calls on a specific extension
On Wed, Nov 22, 2006 at 04:51:26PM -0800, Ira wrote: At 03:14 PM 11/22/2006, you wrote: The missing piece of the puzzle: I'm extension 203. I want any call I park to get parked at extension 2203. I want a call my boss parks to park at 2205, since he's ext. 205. In other words, I want calls parked FROM extension XYZ to be parked AT extension (XYZ+2000). I don't see a way to force parked calls to a specific extension. I'm probably just missing the answer, but I've googled for it and I can't find it. That doesn't seem to be the way parking was designed. It's a first available distribution of a series of numbers you choose. The problem with your plan is that it can't handle a second call on an extension. Coming up in V1.4 is something called SLA or shared line appearance which might do what you want depending upon how it's implemented. For the moment you just need to tell people extension to pick up to retrieve a parked call. Here it's always 701 as we've never yet As I was noting in an earlier message, the parking lot concept is to my view not a thrilling interface at best, and I can't see many times one would want it in a SOHO environment.It seems best for a large PBX where people are moving to random places to pick up calls, and many calls may be parked at any given time. For many people, a far simpler interface is to just put the call on hold -- by pressing just one hold button, and then go pick it up as easily as possible somewhere else.Shared line systems help to do that but from a different direction. The parking lot approach has you remember a somewhat random number told to you, and then to go dial it.People can remember their own extension much more easily, so one good interface in that case is a way to dial a number prefixNNN to pick up a call held on a specific extension (in my pickup group). Or more simply, to dial the pickup number, and if there is only one call on hold, it gives it to you, and if there is more than one, it lets you dial the extension that put it on hold and reads the extensions that have calls on hold to remind you. This is a better interface in an environment were the small security risk here is minimal, such as a home or small office. The nice thing about this interface is that a phone speed-dial function button can be programmed to the pickup number. This means that parking and getting a call can amount to pressing one button to put the call on hold, moving to another phone and pushing another button to get the call, which is about the simplest interface and the one found on key systems and some pbx. Where security is a concern (and the current call parking lot does not actually provide a great deal) you can have a call transferred to a valet, but not require the user to remember a parking lot number if they know the number of the extension that put the call on hold. The valet system gets us partway from what I read, but it still uses the arbitrary number slots. It still requires the user know to transfer a call to the valet. Of course if you know what phone you are going to, you can just do unattended xfer to it, as long as there is not too short a voicemail timeout. But again that's a way more complex interface than push hold and push pickup. ___ --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] Why Aastra uses 48V whereas other IP Phones usemuch less, i.e. 5-12V
On Wed, Nov 22, 2006 at 11:29:01PM -0500, Michelle Dupuis wrote: 48VDC is a long time telco standard - and has become the Power over Ethernet standard. Keep in mind that 'electricity' isn't the measure - it's power. Power is not synonymous with voltage. More to the point, there is a tradeoff. For a given power, the higher the voltage, the lower the current. The lower the current, the thinner the wire you can get away with. Power over ethernet uses very thin wire, so you want high voltage and low current. Power transmission lines use very high voltage because they need (comparatively) low current through the wires. The higher the voltage, the more power you can put through the same wire. To a point. As voltage gets higher, it also gets more dangerous, and needs a bit more insulation. It's very hard to hurt somebody with 12 volts. And 48 volts, while not quite as safe, is still pretty safe. It's been chosen as a voltage that mixes the right combination of safety and power. The higher the voltage, the more heat you can generate if you have the current behind it. (If you are current limited or fuse/breaker protected you are just as safe from fire if things are calibrated right.) In the past, we often drove things with batteries, or wanted to sometimes. Getting 48v with batteries takes a lot of cells with most technologies. Phone central offices had big banks of batteries -- no problem. Today, with advanced switched-mode power supply technology, we can turn just about any voltage into any voltage. So we don't care as much about being able to run on batteries as low voltage, though it's still nice in portable tech. And of course the chips all run on very low voltages today (TTL was 5 volts and it's getting rarer) and they want to be low power.Most of the PoE phones that take 48 volts are converting it down to lower voltages to use. But 48 is a good voltage to be sending on the wires. The USA uses 120v for house current. That's enough to hurt you and can kill you if you touch it wrong, though I've touched it a few times. A lot of the world uses 220. This causes enough of a spark that they require all receptacles to have a switch on them so you don't plug things in live. On the other hand, 220 can deliver twice the power in the same current. Kettles in the 220 world are _really_ fast. Your dryer and oven run on 220 even in the 110 world, only way to get enough power. Same with electric car chargers. ___ --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] Call park on Linksys 922 and similar phones?
I'm having an issue with call park on my new Linksys 922. It has soft menu keys for doing call transfer (which I always think is a good idea because it's amazing how every phone has a different xfer interface and people always get confused). However, I can't get a good call park working on it. It doesn't respond to the use of # for transfer (nor should I want it to, since it has soft transfer keys). If I hit xfer and call 700, the parker does announce the call being parked at 701, but then instead of disconnecting me I hear hold music on the 722 (and continue to hear hold music on the calling phone.) If I hit resume, I am back talking to the calling phone. If I hit xfer again (which is normally how to complete a transfer) both phones disconnect, and the console says that the 922 got tired of parking. --- I must admit, on a side note, I have never been particularly happy with the parking interface. I know a number of other people feel the same since there have been calls and development efforts for ways to improve it, including hints for BLF, shared/bridged line functionality etc. For the SOHO application, ie. a home pbx, the idea of a parking lot with numbered slots is generally overkill. Such a home is extremely unlikely to ever have more than one call parked in a pickup group, or per PBX frankly. I think a much nicer interface would be to have the first phone simply put the call on hold (which is the typical approach in many key systems) and then dial an extension to pick up the call that's on hold in my pickup group. If, as will rarely be the case, more than one call is on hold, I think the best way to deal with it would be to present an IVR that says: 3 Calls are on hold. Please enter the extension that placed the call on hold. Available extensions are 305, 49 and 902. But 99 times out of 100, the interface would amount to putting the call on hold, going to another phone and hitting the pick up held call speed dial. Which is what people tend to like in SOHO settings. You can sort of do this if you just insist there is only one parking slot, but it won't handle the rare double-hold case and it's much more to do when putting the call on hold. Any effort been made in this direction? ___ --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] Powering SNOM 200 phones?
A follow up on my message about my SNOM 200 phones now powering from my 802.3af Netgear FS108p PoE box. To follow up for those finding this thread on searches... I purchased some PowerDSine 6001 units (very cheap on ebay) and they power the SNOM 200 fine. Some Buffalo units also did this. So it seems that either the Netgear is too picky about its detection, or the SNOM 200 not fully compliant. The powerdsines are big and require an extra cable as all external injectors will, but they work. ___ --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] Powering SNOM 200 phones?
Ok, not exactly an Asterisk problem, but... I picked up some SNOM 200 phones because SNOM's have been recommended for use with Asterisk and they have line buttons that can subscribe to presence. However, they don't appear to power up when connected to my Negear FS108P, which is an 802.3af Power-over-ethernet capable hub. I am pretty sure these are the SNOM 200b, in that the ethernet connectors are at the back rather than on the bottom, and there doesn't even seem to be a jack for plugging in any other kind of power adapter (and I don't have another one.) Anybody had experience with these phones and powering them? Is it just an icompatability with the Netgear, or do I have 2 dead phones? Would getting a different PoE box be a good idea? (Frys has the airlink for $29 from time to time, which is a great price. Otherwise many older PoE boxes tend to cost more than the modern cheaper phones they might power.) ___ --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] Powering SNOM 200 phones?
On Fri, Nov 10, 2006 at 02:34:31PM +1100, Paul Hales wrote: I had a 200, and it worked fine with POE. The standard power connector was the RJ-11 style as mentioned below. Weird item that one. The successor to the 200, known as a 190 does NOT support poe, while the 320 does. Yeah, these have an extra unmarked rj-11 on the bottom next to two covered holes (with nothing but pc board behind) where the ethernet would be on the old model of snom 200 if I read the manual right. So that's the power. So I guess the only way to find out if they just don't talk to my netgear POE (which does power my grandstream 2000) is to find different POEs. Or buy the power supplies which don't seem to be very expensive -- or are there different models of snom power supplies? It is suggested the 190 takes 5v, not 48v. ___ --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] Java Web Phone
On Thu, Nov 02, 2006 at 11:23:08AM -0500, Guillermo Salas M. wrote: On Wed, 2006-11-01 at 16:05 -0500, Vladimir Montealegre Estailes wrote: Hello list partners you know about a softphone made in java attachable in a web page? GNU! I'm using JIAXClient [1] to permit to any user to join one meetme room [2] with the IAX2 protocol, works very great for me, and is very easy to install and modify to your needs. [1] http://www.hem.za.org/jiaxclient/ [2] http://www.rmsenecuador.info/jiaxclient/index.html Useful, but it requires a signed client with permissions to install a DLL. Once you are able to do that, you can do anything -- install a full softphone, send the voice to any destination (not just back to the server) -- and of course take over the other person's machine so they should be very wary of approving the client. What would be more useful is an applet that can run as a pure applet. That's forced to only talk to the server that served it, but it doesn't need approval. Or, in theory the voip client going into the new flashplayer. ___ --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: Asterisk both behind a NAT and outside at the same time
On Wed, Nov 01, 2006 at 06:16:25PM +0100, Benny Amorsen wrote: BT == Brad Templeton [EMAIL PROTECTED] writes: BT The correct behaviour, as I see it is: BT a) Native bridge when connecting two external channels -- BT everybody is on the real internet b) Native bridge when connecting BT two internal channels -- everybody is on the 192.168.* network c) BT Route RTP through Asterisk when connecting internal and external BT d) When a channel is to a device behind a remote NAT, the usual BT rules apply (either use STUN or other smart NAT, or route RTP BT through Asterisk) You won't get asterisk to do what you want. That kind of logic simply isn't implemented, and no amount of fiddling with configuration files will make it happen. I'm sure patches are welcome. Thanks. Will look into it. Probably need to switch to 1.4 before I start writing more patches though. Though to my surprise I am now discovering something worse. It doesn't seem to work in the lastest 1.2 even with canreinvite=no and nat=yes on the natted (internal) phone with a connection coming in from outside.The outsider has to presume it's calling a natted phone rather than a non-natted asterisk, the invalid SDP is leaking out. I'll see if I can pin that down a bit better. ___ --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] Asterisk both behind a NAT and outside at the same time
I've read a lot of the descriptions of handling NAT with Asterisk, and the use of both the nat and canreinvite flags. I am very familiar with Sip and NAT but have not seen an answer to the following question. My Asterisk server runs on a machine with two ethernets. One is an external net, with exposed IP addresses. The other is an internal net with natted IP addresses. Thus the server has two addresses. The server is _not_ the NAT gateway. That's a linksys box which has its own external IP to gateway traffic from the internal natwork. The phones are on the internal NATwork. Asterisk talks to them over it. Outside peers, such as SIP termination providers etc. talk to the Asterisk server via its outside address, which is as you would expect. However, from time to time I get the famous one-way audio because Asterisk has decided to do a native bridge between a natted SIP phone and an external SIP peer. It sends the internal IP of the SIP phone in the SDP and of course the outside service can't send packets to that. I could just turn off reinvites on the internal phones, but this would cause them to route all traffic through the asterisk box, even on internal calls between phones on the same ethernet, which seems foolish to me. I don't want to turn off reinvites to the external peers -- if a call comes in from a SIP originator for example, and is send back out to a SIP terminator (call forwarding) I want a native bridge for sure.(Handling the internal traffic is not so much of a burden though sometimes I hear latency because of it, but routing external traffic through the asterisk box is a bad thing.) So what I want is for Asterisk to use native bridges when connecting two channels behind the NAT, or two channels on the real internet, but not to do so when connecting an internal and external channel. It should be able to see the IP addresses, and know the difference between natted and external ones and know they can't talk to one another. (The ICE protocol would handle this someday.) Is IAX smarter about this? Of course I might even want to get smarter about this. Is it possible, typically by configuring stun in the phones, to have them be aware of their external IP and tell Asterisk about it? With a full cone NAT, it would work to do a native bridge between the internal and external devices so long as the external device is given the right address and port of the NAT box, not the internal address of the phone. However, we don't want to do this on internal to internal calls -- many NATs can't hairpin. I would think this would be a common situation (though perhaps more commonly the asterisk server IS the firewall/NAT.) Is there a solution that does the right thing most of the time? ___ --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] Asterisk both behind a NAT and outside at the same time
On Tue, Oct 31, 2006 at 07:40:35PM +0800, Leo Ann Boon wrote: Have you tried setting the externalip and localnet parameters? Localnet makes some sense, and is set (should be the default anyway, no?) externalip, as I understand it, is for an Asterisk which is behind a NAT. This asterisk is not behind a NAT to anybody. The phones are behind a NAT to the outside world but not to the Asterisk box, which has two ethernets on it, one for the internal natwork and one for the real internet. It uses bindaddr=0.0.0.0 and listens to both addresses. Sorry for my previous post I misunderstood the problem. You should set canreinvite=no to all sip peers that connect from outside. That's precisely what I don't want to do. This would block native bridging in the one case where it's most important. The correct behaviour, as I see it is: a) Native bridge when connecting two external channels -- everybody is on the real internet b) Native bridge when connecting two internal channels -- everybody is on the 192.168.* network c) Route RTP through Asterisk when connecting internal and external d) When a channel is to a device behind a remote NAT, the usual rules apply (either use STUN or other smart NAT, or route RTP through Asterisk) The super correct behaviour, which I don't expect but would be nice is e) Clever native bridge between internal and external by being aware that the device talks to the outside world using a different address than it talks to you. (Possibly if the phones use STUN they will tell Asterisk their external IP, which is not the same as Asterisk's though it's on the same subnet) I have used localnet=192.168.* and nat=yes on a local device and it still attempts an incorrect native bridge between internal and external, with one-way audio. If I do canreinvite=no on the local devices then it works of course, but now means the local phones will never native bridge amongst themselves. In a larger network, that would be a problem, and it's a poor result in any network. This is the latest svn of 1.2, by the way. ___ --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] Asterisk both behind a NAT and outside at the same time
On Tue, Oct 31, 2006 at 04:51:44PM -0500, C F wrote: The correct behaviour, as I see it is: a) Native bridge when connecting two external channels -- everybody is on the real internet It might not work if one of them is NATed. Therefore the correct way to do this is to use canreinvite=no Of course, if the external peer or user is natted, you would want to turn on nat and canreinvite=no for such channels. However, I do not wish to set canreinvite=no for an external peer like another asterisk box with an external IP, or a SIP termination provider. More than do not wish -- this is the most important case. b) Native bridge when connecting two internal channels -- everybody is on the 192.168.* network canreinvite=yes will take care of this. Of course, but the point is that the internal channels (local SIP phones) are involved in connections to both local phones, and to external peers. c) Route RTP through Asterisk when connecting internal and external Again by adding canreinvite=no to externals you have this. But this defeats the entire purpose. Here are two situations where you would most definitely not want to have canreinvite=no a) Call comes in via SIP origination, and you direct it back out to a PSTN phone via your SIP termination. You want the RTP to go directly from the originating point to the termination point, not to hairpin through your asterisk box, which would just add latency and eat bandwidth. b) Click to call APP I have where I connect two PSTN endpoints. Again, it's necessary not to hairpin the RTP. c) Double all this with some advanced providers who, once they figure where the call is actually being terminated, do their own native bridging and direct your RTP to the actual PSTN entry point. There it's possible to get the RTP to go by the shortest (and lowest latency) path it can. But not if you hairpin it. d) When a channel is to a device behind a remote NAT, the usual rules apply (either use STUN or other smart NAT, or route RTP through Asterisk) How will asterisk know? The correct *setting* (not behavior) is canreinvite=no for the external devices. I would have to differ. That's the right setting for external user devices behind NAT. Do you believe it's correct for devices not behind NAT? Asterisk can tell if a device is behind NAT if the device has been made in the last few years, because such devices support a variety of techniques to inform the server they call that they are behind NAT, and even what their external IP is if need be. However, reinvite is not safe with a symmetric nat unless there is really good cooperation. So I understand turning off reinvite for any external device behind NAT. (Of course a clever box can notice that it sees two devices that have NAT addresses on the same subnet and both are using the same external IP. In that case, it can tell them to send their RTP directly to one another which is very much the best thing to do. This allows things like a branch office using a head office Asterisk server and calls within the branch staying on the LAN. This is what the ICE protocol is supposed to solve, of course.) Why are you so against having the RTP go thru asterisk? For external connections? There are a ton of reasons, some outlined above. An asterisk box with proper use of native bridging can handle a virtually unlimited number of calls.Put the RTP through it and it can handle only a modest number, and reduces the quality of those calls significantly. Having internal calls go through the asterisk box is not as much of a problem, but I have noticed latencies because of it even on my internal LAN, though I have not pieced together exactly why, it's almost certainly because of the RTP bridging. Which I always get when I call IAX to SIP of course. It's not because of load. Anyway, my main question is, has anybody figured out how to make Asterisk do the right thing here. I am surprised if my configuration is that unusual. Having both an internal and external network is pretty common at a lot of places. And a server that's on both is, I think, quite common, so I had not expected this to be a difficult thing to figure out how to do. ___ --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] Opinions on the best wholesale origination/term providers
I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In particular, this means they probably tie into the PSTN at multiple points, definitely east and west coast and also in Europe. I don't want a California caller calling California to have to send their packets to the east coast and back. (This made me discard RNKVoIP, which was high on my list) b) Fair pricing. I've seen blended rates down to a penny, and non-blended down to half/cent in the big city Tier-1s. I don't expect the lowest possible price but I don't want to see 100% markup either. For a blended rate, let's see under 1.5 cents to the USA and Canada. (Canada is actually down to .8 cents at some providers now, others charge more for it.) c) Origination, also at a fair price Which seems to be about $1/month for DID in USA, $2 in Canada, and close to 1 cent/minute. But I can pay more to get other factors. I guess I can go to another firm for origination outside the USA in a pinch. d) Reliability very high. Duh. e) Decent customer service. If things go down you fix them and I can reach you to fix them. I don't need handholding, I know my tools, but I do need you to fix problems. If you know your Asterisk, linux and SIP even better. f) Decent automated interface. So I can get DIDs, configure IPs, billing etc. g) Static IP authentication It's faster. Though dynamic IP registration as a backup is handy. h) Global termination I don't want to have to manage and support too many different providers. That's work for me. So give me good global termination prices too. That knocked out termination.com/icall Though if I can't get all I want, I guess I'll buy global from one company and domestic from another. i) No high minimums I am just testing my software apps right now so I'm not going to bill minutes until much later when they ship. So I can't give you tons of minutes per month. I don't mind prepaying. j) SIP, and decently implemented. Asterisk/SER is fine. Now we get to my nice to have list o) IAX as well as SIP. Makes testing stuff easier. o) DTMF via SIP-INFO. This lets me have native bridge for the voice but still hear the DTMFs at my server, which would be handy. o) Origination worldwide o) Toll free origination o) Cheap toll free termination. (Why does this cost money anyway?) o) Don't want E911 service now. Might want it in future. Don't want to pay now. So here's what I have found that come close sellvoip -- good quality, low latency, good price. Online tools suck, customer service nonexistent rnkvoip -- most of what I want but east coast gateways only. Good customer service but som unreliability in equipment telcommone.net -- Looks fairy good so far. $2/DID in small quantities, but comes down eventually. Very good term prices. Claims to enforce instate calling prices. (Old world thinking) termination.com -- very good prices but USA only terravon -- 1.7 / minute. trxtelecom -- offers free 800 termination, they claim, and pay-you origination in rural latas if that's your style. (Great if you expect most calls to come from cell phones or other people with bundled long distance blended rates.) unlimitel -- for canada netiqsys.net -- no origination but good prices Any views on these or other providers? ___ --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] Asterisk both behind a NAT and outside at the same time
On Wed, Nov 01, 2006 at 08:10:29AM +0800, Leo Ann Boon wrote: Brad Templeton wrote: The way I understand it, externalip and localnet work hand-in-hand. I do agree with you that this is commonly used for Asterisk behind a NAT. I believe these parameter just helps asterisk determine what to do. In your case, you don't lose anything - the external IP would still have to be written into every outbound packet. It's called externip I think, not externalip. I have set both externip to be my external IP address, and localnet to be the natwork, and even set canreinvite=no and nat=yes and the SDP I get back from an invite to [EMAIL PROTECTED] still has 192.186.* in it. It uses bindaddr=0.0.0.0 and listens to both addresses. externalip doesn't affect the bindaddr. Would not expect it to. Just trying to be clear to people that the machine has two ethernets. I was hoping Asterisk would just automatically say, Wait a minute, I'm taking an SDP with addresses in the localnet, and sending it out to a peer on the outside internet. That's not going to work! Now one of my tests has a SIP program I have written attempt to call Asterisk. It sits on port 5061 invites to Asterisk on 5160 of the machine with the external address as follows: INVITE sip:[EMAIL PROTECTED]:5160;transport=udp SIP/2.0^M Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE^M From: Voxable sip:[EMAIL PROTECTED];tag=3445^M To: Party Leg1 sip:[EMAIL PROTECTED]:5160^M Via: SIP/2.0/UDP 198.144.201.82:5061;branch=z9hG4bK6563eba5fd430b5af93579617a44450e^M Max-Forwards: 12^M Contact: Caller App sip:[EMAIL PROTECTED]:5061^M Date: Wed, 01 Nov 2006 04:52:07 GMT^M User-Agent: Voxable 0.1^M Content-Type: application/sdp^M Content-Length: 154^M ^M v=0 o=capp 1162356727 1 IN IP4 198.144.201.82 s=CApp3PCC c=IN IP4 198.144.201.82 t=0 0 m=audio 5308 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PMCA/8000 Asterisk sends this on to the phone, but rewrites the SDP to present a local one: o=root 26391 26391 IN IP4 192.168.123.10 s=session c=IN IP4 192.168.123.10 t=0 0 m=audio 10856 RTP/AVP 0 97 8 101 I answer the phone and it responds to this SDP with an OK o=brad 8000 8000 IN IP4 192.168.123.18 s=SIP Call c=IN IP4 192.168.123.18 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 Asterisk then sends back this SDP without rewriting it. Is it only doing that because it knows the traffic came from the same machine? Asterisk forwards this OK back. Note the SDP SIP/2.0 200 OK^M Via: SIP/2.0/UDP 198.144.201.82:5061;branch=z9hG4bK94e379b15dbd22f8594fa6e88a4cfcc0;received=198.144.201.82^M From: Voxable sip:[EMAIL PROTECTED];tag=3445^M To: Party Leg1 sip:[EMAIL PROTECTED]:5160;tag=as62de8d32^M Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE^M User-Agent: Caller Asterisk^M Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY^M Supported: replaces^M Contact: sip:[EMAIL PROTECTED]:5160^M Content-Type: application/sdp^M Content-Length: 199^M ^M v=0^M o=root 26391 26391 IN IP4 192.168.123.18^M s=session^M c=IN IP4 192.168.123.18^M t=0 0^M m=audio 5004 RTP/AVP 0 8^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=silenceSupp:off - - - -^M a=sendrecv^M My software then forwards that SDP on to an outside location, where the SDP is useless. It works if the outside provider I forward the SDP to has my asterisk box set with some flags (nat=yes I presume?) though I can't figure why. That box is presumably, seeing the internal address, routing the audio to some port on the * box, and asterisk is forwarding it but I can't see how this is happening. Odd. ___ --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: Opinions on the best wholesale origination/term providers
On Tue, Oct 31, 2006 at 08:03:56PM -0800, Martin Joseph wrote: On 2006-10-31 17:29:47 -0800, Brad Templeton [EMAIL PROTECTED] said: I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I also use Sellvoip and I am close to them (Seattle). They by FAR produce the best call quality for me, when compared to nufone and Teliax, although both of those companies do ok, my routes to them aren't nearly as clean. I recommend Teliax for good support. Their DIDs ($5/month plus 2 cents/minute) are much too high, their termination is 2 cents which is tolerable but in general too high for a wholesale service. But thanks for the comment. The sellvoip guys (guy?) are indeed producing good quality. Another thing they are doing, which I really like, is processing termination quickly, in that when I do the invite it's ringing within a fraction of a second. A few other termination providers I have tried are taking 3-4 seconds to ring after invite. You thought I wrote a lot and I didn't even put that on my list. We just have to convince Jed at Sellvoip to hire some some support techs, even if he has to add a couple of tenths per minute. ___ --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
