Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)
Marty, Where are you paying $1000 for a 1600 series Cisco? I can get you 20% off that price on any quantity (note: Sarcasam). Its not the 1990's anymore. You can get them on eBay ($50-150) for only slightly more then the Linksys. The performance is rock solid. Three-quarters of the world have used them for decades. I know of units running 2 and 3 YEARS between reboots. The power company reboots my equipment more then I do. Ok it is true that Cisco does not support the models anymore, but you can't buy a services contract for a linksys router either. It can sometimes be a little difficult to configure without any technical knowledge but that is what most of us get paid for. It does impress the customer when you bring in the grey box labled Cisco. As for performance just try to put 50 people behind a linksys/netgear/dlink. I've used 1605R supporting +100 users. Not even a blink. Finally, untill everyone is using 10Mps FTTH the broad band link is still the slowest part of the connection. Not to shabby for antiquated technology. Mark C Martin Joseph wrote: On 2007-01-06 00:48:11 -0800, Mark Coccimiglio [EMAIL PROTECTED] said: Mike I'm using a Cisco 1605R [running IOS 12.3(5a)] small office router with Fair-Weight queueing enabled. Works great. The nice thing about Fair-Weight queueing is that it dynamically adapts to lower the priority of higher demand traffic (e.g. large downloads). If you want quality stick with quality stuff. Mark C Reread the subject line please. $1000 (US) isn't inexpensive by any stretch. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/TCP?
On 5 Jan 2007, at 23:22, Yuan LIU wrote: From: James R. Stevens [EMAIL PROTECTED] TCP is a connection oriented protocol ..as others mentioned, it superiority comes because it knows when packets are dropped to resend them. It also has mechanisms for flow control etc.. SIP is a connection-less protocol. It uses 'best effort' transmissions..if u want its delivery guaranteed you must encapsulate it. So I take it that UDP is just a decision due to popular demand; timing (jitter) is a frequently cited factor to favour UDP. Is there any technical difficulties in implementing SIP/TCP within Asterisk? The reason I'm asking is that there are products that support both UDP and TCP. And SIP/TCP, RTP/TCP have their own merits. Granted, SIP is connectionless. So is HTTP (well, for its original design anyway). I notice that guaranteed delivery could be a good thing for SIP in many situations; there have also been advancements that make timing less an issue in RTP/TCP. Is switching to SIP/TCP - RTP/TCP as simple as rewrap messages/ streams, or is it more involved? It's a latency thing. Say you send a packet every 20ms Say you have a pair of endpoints with 100ms between them. Say you drop a packet in the media channel. TCP will re-request the missing packet, which has 2 bad effects: 1) when it does turn up the packet will be 100ms late so you have to play silence or make something up until it does. 2) all the subsequent packets will be behind the re-tried packet and also 100ms late. - Note that because TCP is a stream, you can't get at the subsequent packets even if they turn up on time because you have to wait for the missing one. Even more frustratingly you now have to dump these 4 perfectly good packets. If you don't you will have introduced 100ms of lag in the audio stream. - Of course none of this applies if you are on a LAN - with 1ms roundtrip time as the retry can get to you soon enough to be useful. With UDP you simply make something up for the missing packet and carry on when you get the next one. - so you make up a single packet instead of 5. With TCP the lost packet is multiplied by the ratio of the roundtrip time to the packet interval. Of course you can cover this up by increasing the buffering, but then you are introducing yet more lag. So, I simply don't think that TCP is suitable for telephony media streams over any network where the roundtrip time is of the same order as the packet interval. Now there are 'reliable datagram protocols' ( IL for example) but they aren't much used on the internet. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] no unicall on 1.4
On 6 Jan 2007, at 19:37, Anton Krall wrote: Anyway, we are drifting from the initial point which was to hope and support further development of R2MFC on the asterisk community so I propose a bounty to get Steve (sheesh, that's sounded like a hit bounty :)) I meant, a bounty to convince Steve to help the community that still uses R2 what do you say? I'm not sure that Steve is likely to be bribed into contributing I didn't get the impression that it was a matter of money, more to do with commit access and technical design. Put a bounty up by all means, but leave it open to anyone who has the time/skills to maintain the unicall- asterisk channel driver - or perhaps write a new one. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] Scalable IVR with asterisk
snacktime wrote: I'm looking at a project that is basically just an IVR, but will potentially be handling 20K or so calls per day, maybe even more. Any reason why asterisk would not work for this? I'm thinking in terms of being able to distribute calls across an asterisk server farm from some type of central termination/switching/proxy hardware, whatever that might be (Max TNT, etc..) I can't think of any inherent reason why this won't work. Anyone else? Chris I use it with no problems in a 12,000 calls a day call center environment with no problems. IVR should be be no problem. Thanks, Steve ___ --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 get dial tone back
Am Samstag, den 06.01.2007, 23:19 -0800 schrieb Yuan LIU: After the user navigated some voice menus, how do I give him another (fake) dial tone? If you want the user to get the tone meaning please dial a number now, perhaps DISA is the right you. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA BR Anselm ___ --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] Reserved extensions?
Am Samstag, den 06.01.2007, 23:02 -0800 schrieb Yuan LIU: I'm creating extensions _*XX. But whenever I press *0 or *8, Asterisk throws out congestion and hangs up. I set verbose to 6 and debug to 6, but all Asterisk cares to display in console is -- Starting simple switch on 'Zap/1-1' -- Hungup 'Zap/1-1' Are these combinations forbidden? Check features.conf - there are combinations reserved for transfer etc. In my feature.conf, *0 was reserved for hangup (until I commented it out). My personal opinion for dialplans is that they should not use the * and # keys for regular numbers - it just does not feel right. You are of course free to differ :-) BR Anselm ___ --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] Dimensioning a 50 sip phone installation
On 7 Jan 2007, at 07:28, Erick Perez wrote: The customer found a VIA EPIA 1.2ghz with 1gb ram, one pci slot mini.itx. Can this equipment handle a sangoma/digium E1 card with 25 SIP ulaw phones +voicemail and *no* call recording? Yes. I've got the 1ghz version and it is fine (even doing 5 channels of alaw - g729a) BUT I'm not entirely happy with these motherboards. I've just had 2 (out of a total of 5!) die. In both cases the capacitors were faulty. The machines were 1 year old and out of warranty, so I couldn't get replacements foc despite the fact that the caps were clearly sub-standard. When I replace these systems I won't be buying VIA EPIA again. If you do buy them, make sure you spec a CD drive in the package. I didn't and OS rebuilds are a pain to do via a USB DVD drive. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] Reserved extensions?
Yuan LIU wrote: I'm creating extensions _*XX. But whenever I press *0 or *8, Asterisk throws out congestion and hangs up. I set verbose to 6 and debug to 6, but all Asterisk cares to display in console is -- Starting simple switch on 'Zap/1-1' -- Hungup 'Zap/1-1' Are these combinations forbidden? Yuan Liu Check your features.conf http://www.voip-info.org/wiki-Asterisk+config+features.conf Thanks, Steve ___ --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] fax transmission
Doug Lytle wrote: Vieri Di Paola wrote: advantage of sending faxes through Asterisk with, say a 8-port ISDN card is that I don't need any modems and I have a lot more channels to work with. Less maintenance, too. Or maybe I'm missing something to do the same with iaxmodem is the component that will allow you to use all of your 8 ports on the ISDN card as software modems. It works very well. It is worth noting that it works best if the asterisk server with the linecard is the same machine as the hylafax server. (although afaik most problems with remote servers are now fixed in the latest release). ___ --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] LUSYN patches
Hello all, New versions of the history buffer patches for v1.4 of Asterisk and Zaptel are now available at www.lusyn.com/resources/asterisk/usehist.htm. The patches for v1.2 will remain available for a few months. The ring begin patches are now obsolete, as the history buffer patches can be used as a workaround. Rgds, Marc ___ --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] Best inexpensive home office router for VoIP (QoS with maybe PoE)
We've had a lot of success with Thompson Speedtouch 780 routers, which have built in adsl modems, and two ATAs. They don't seem to use QoS in the strictest sense, but do a very good job of prioritising the traffic from their own ATAs. If you're happy to stick with analogue hansets instead of the SIP hardphones, they provide an excellent protection to upload bandwidth. They also seem to do some early dropping on incoming traffic to persuade the ISP's routers to slow down downloads once a call has been going for a bit, hence they can limit downloads as well. simon On 4 Jan 2007, at 17:56, Mike wrote: Hi, I'm looking for opinions on the best value router to use for home offices. It should work for a scenario in which there are 3 computers and 2 SIP phones, handling QoS so that the phones always have higher priority traffic than the PCs. (and not rely on the phones to do the QoS because some PCs may not be connected to the phones). QoS could be based on destination and source IP (i.e. an Asterisk server) or MAC address of the phones. Ideally with PoE, but at this point it's just a bonus. What are people on this list using? I've found that the mention QoS on a box doesn't guarantee any real QoS functionality. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which is GUI to edit Asterisk IVR logic
Olivier wrote: By Trixbox, do you mean FreePBX (formely AMP) ? Yes. Sorry, I tend to group them together and should have noted that they are separate products. -- Warm Regards, Lee ___ --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
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) 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. ___ --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
Unless bandwidth between the * servers is a concern, then you're better off keeping the link between servers as IAX. (preferably trunked) As I understand it video will NOT work if you use an IAX trunk between * boxes, it must be SIP. Just food for thought in case you are planning on using video. David ___ --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] fax transmission
Thomas Kenyon wrote: Doug Lytle wrote: iaxmodem is the component that will allow you to use all of your 8 ports on the ISDN card as software modems. It works very well. It is worth noting that it works best if the asterisk server with the linecard is the same machine as the hylafax server. (although afaik most problems with remote servers are now fixed in the latest release). Unfortunately, the problem with using iaxmodem remotely from the Asterisk server (jitter disrupting the audio) is not something that can be fixed from the software-side of things. The 0.1.16 release included a type of fixed jitterbuffer that helps iaxmodem prevent itself from mistaking jitter as carrier loss when there still should be carrier or vice-versa (carrier when there should not be). This fixed jitterbuffer was not working right until the 0.2.0 release, however. So this jitterbuffer cannot recover the disrupted audio, although it does help mitigate the effects of it. The only true fix must come by eliminating the causes of jitter - and that usually means placing iaxmodem on the Asterisk system. The iaxmodem documentation does discuss this in detail. Thanks, Lee. ___ --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/TCP?
SIP over TCP != RTP over TCP The whole latency deal is much more of a concern in RTP (as well as trying to deliver a late packet, that will be not very useful also). As I understand, MS does SIP/TCP on their LCS or something like that. Still, not RTP over TCP, since it does not make sense for the voice-path. Tim Panton wrote: On 5 Jan 2007, at 23:22, Yuan LIU wrote: From: James R. Stevens [EMAIL PROTECTED] TCP is a connection oriented protocol ..as others mentioned, it superiority comes because it knows when packets are dropped to resend them. It also has mechanisms for flow control etc.. SIP is a connection-less protocol. It uses 'best effort' transmissions..if u want its delivery guaranteed you must encapsulate it. So I take it that UDP is just a decision due to popular demand; timing (jitter) is a frequently cited factor to favour UDP. Is there any technical difficulties in implementing SIP/TCP within Asterisk? The reason I'm asking is that there are products that support both UDP and TCP. And SIP/TCP, RTP/TCP have their own merits. Granted, SIP is connectionless. So is HTTP (well, for its original design anyway). I notice that guaranteed delivery could be a good thing for SIP in many situations; there have also been advancements that make timing less an issue in RTP/TCP. Is switching to SIP/TCP - RTP/TCP as simple as rewrap messages/streams, or is it more involved? It's a latency thing. Say you send a packet every 20ms Say you have a pair of endpoints with 100ms between them. Say you drop a packet in the media channel. TCP will re-request the missing packet, which has 2 bad effects: 1) when it does turn up the packet will be 100ms late so you have to play silence or make something up until it does. 2) all the subsequent packets will be behind the re-tried packet and also 100ms late. - Note that because TCP is a stream, you can't get at the subsequent packets even if they turn up on time because you have to wait for the missing one. Even more frustratingly you now have to dump these 4 perfectly good packets. If you don't you will have introduced 100ms of lag in the audio stream. - Of course none of this applies if you are on a LAN - with 1ms roundtrip time as the retry can get to you soon enough to be useful. With UDP you simply make something up for the missing packet and carry on when you get the next one. - so you make up a single packet instead of 5. With TCP the lost packet is multiplied by the ratio of the roundtrip time to the packet interval. Of course you can cover this up by increasing the buffering, but then you are introducing yet more lag. So, I simply don't think that TCP is suitable for telephony media streams over any network where the roundtrip time is of the same order as the packet interval. Now there are 'reliable datagram protocols' ( IL for example) but they aren't much used on the internet. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax transmission
--- Doug Lytle [EMAIL PROTECTED] wrote: iaxmodem is the component that will allow you to use all of your 8 ports on the ISDN card as software modems. It works very well. Thanks Doug. As you can see I'm new to iaxmodem... will try it out. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get dial tone back
On 1/7/07, Yuan LIU [EMAIL PROTECTED] wrote: After the user navigated some voice menus, how do I give him another (fake) dial tone? Have you try this? from http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Playtones Example 2 Especially useful for the s extension: exten = s,1,Answer exten = s,2,Playtones(dial) ;use DigitTimeout previous to Asterisk 1.2 exten = s,3,Set(TIMEOUT(digit)=5) exten = s,4,WaitExten(60) atik ___ --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] Parking a call a second time using #700..
I wouldn't address your mailing to [EMAIL PROTECTED] .. seems you've written .conf way too many times. :) On 1/6/07, Marc Archer [EMAIL PROTECTED] wrote: Hi all, Is there anyway you get Asterisk to let you park a call a second time via pressing the # key + 700 (or whatever is in features.conf) ? If you get a call out of the parking lot hit # again it just sends the DTMF tone down the line and nothing happens.. A few posts have hinted that you cannot blind transfer a parked call, but surely there must be some way around this as I can get the call to park by again using the orbit function key on our snom phones (but I'd like to be able to hear the announcement for which lot it goes into). Marc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get dial tone back
From: atik khan [EMAIL PROTECTED] On 1/7/07, Yuan LIU [EMAIL PROTECTED] wrote: After the user navigated some voice menus, how do I give him another (fake) dial tone? Have you try this? Thanks for the reference and example, atik. Seems exactly what I need. Yuan Liu from http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Playtones Example 2 Especially useful for the s extension: exten = s,1,Answer exten = s,2,Playtones(dial) ;use DigitTimeout previous to Asterisk 1.2 exten = s,3,Set(TIMEOUT(digit)=5) exten = s,4,WaitExten(60) atik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get dial tone back
From: Anselm Martin Hoffmeister [EMAIL PROTECTED] Am Samstag, den 06.01.2007, 23:19 -0800 schrieb Yuan LIU: After the user navigated some voice menus, how do I give him another (fake) dial tone? If you want the user to get the tone meaning please dial a number now, perhaps DISA is the right you. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA BR Anselm Many thanks, Anselm. I have overlooked DISA - I have another use case that could use it. So I take it that there's a DISA context global variable? This is cool! Yuan Liu ___ --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] Reserved extensions?
From: Anselm Martin Hoffmeister [EMAIL PROTECTED] Am Samstag, den 06.01.2007, 23:02 -0800 schrieb Yuan LIU: I'm creating extensions _*XX. But whenever I press *0 or *8, Asterisk throws out congestion and hangs up. I set verbose to 6 and debug to 6, but all Asterisk cares to display in console is -- Starting simple switch on 'Zap/1-1' -- Hungup 'Zap/1-1' Are these combinations forbidden? Check features.conf - there are combinations reserved for transfer etc. In my feature.conf, *0 was reserved for hangup (until I commented it out). Got it. Thanks! These combos are themselves features, only implemented as dialplans. So maybe I should learn how to properly create new features. Yuan Liu My personal opinion for dialplans is that they should not use the * and # keys for regular numbers - it just does not feel right. You are of course free to differ :-) BR Anselm ___ --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] Which is GUI to edit Asterisk IVR logic
On Fri, 5 Jan 2007, Olivier wrote: By Trixbox, do you mean FreePBX (formely AMP) ? Trixbox includes FreePBX but it also includes some other stuff like FOP and an open-source install of SugarCRM. FreePBX is also available as a standalone program. -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --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] no unicall on 1.4
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Eric ManxPower Wieling wrote: Anton Krall wrote: This is exactly one of the things that Steve and I discussed a bit ago... when did asterisk turn from an open source project with very good developers into a business that only focuses in $$$? I imagine that happened around the time they sold their soul to the venture capitalists. 8-) Oddly, I download and install Asterisk for free all the time. Oh, you must be using the warez version then. 31337 then aren't you! :D - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoWJbS6d5vy0jeVcRArCpAJ9nJUq1NHzN/X8DrCMe7yB8LtNXkwCcCRfj 2KojUWrXmmJ/x55GMwvYZoI= =tUpw -END PGP SIGNATURE- ___ --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] Call connected, cannot hear or speak - $20 for fix
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 zero massive wrote: I am able to get this script to dial, but I am unable to talk or hear anything. The script asks for the number to call and the the caller id to display (if user is not at their normal extension). Once submitted, the external extension receives a call, once answered the call is then placed to the dentition number. The script works as the call is place, but I cannot hear or say anything. Any one that is able to get this going I would be will to give $20 to (via paypal) Does it work when you call from one of the phones to the other? Say in the script you are trying to connect User01 with User02. If you make a normal call between these users, is the audio passed? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoWNCS6d5vy0jeVcRAoORAJ4zd5etQcQqntLSdxTWaCzqMwF78ACfW4Jz to2/ubpJXIU+7mQSVfvIIIM= =e0/A -END PGP SIGNATURE- ___ --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] Call connected, cannot hear or speak - $20 for fix
Yes, it seems to fail when both extensions are external On 1/7/07, Matt Riddell (NZ) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 zero massive wrote: I am able to get this script to dial, but I am unable to talk or hear anything. The script asks for the number to call and the the caller id to display (if user is not at their normal extension). Once submitted, the external extension receives a call, once answered the call is then placed to the dentition number. The script works as the call is place, but I cannot hear or say anything. Any one that is able to get this going I would be will to give $20 to (via paypal) Does it work when you call from one of the phones to the other? Say in the script you are trying to connect User01 with User02. If you make a normal call between these users, is the audio passed? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoWNCS6d5vy0jeVcRAoORAJ4zd5etQcQqntLSdxTWaCzqMwF78ACfW4Jz to2/ubpJXIU+7mQSVfvIIIM= =e0/A -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] no unicall on 1.4
Hahahahaha |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Matt Riddell (NZ) |Sent: Sunday, January 07, 2007 3:13 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] no unicall on 1.4 | |-BEGIN PGP SIGNED MESSAGE- |Hash: SHA1 | |Eric ManxPower Wieling wrote: | Anton Krall wrote: | This is exactly one of the things that Steve and I discussed a bit ago... | when did asterisk turn from an open source project with very good | developers | into a business that only focuses in $$$? | | I imagine that happened around the time they sold their soul to the | venture capitalists. 8-) | | Oddly, I download and install Asterisk for free all the time. | |Oh, you must be using the warez version then. | |31337 then aren't you! | |:D | |- -- |Cheers, | |Matt Riddell |___ | |http://www.sineapps.com/news.php (Daily Asterisk News - html) |http://wap.sineapps.com (Daily Asterisk News for your cellphone) |http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) |-BEGIN PGP SIGNATURE- |Version: GnuPG v1.4.2 (MingW32) |Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org | |iD8DBQFFoWJbS6d5vy0jeVcRArCpAJ9nJUq1NHzN/X8DrCMe7yB8LtNXkwCcCRfj |2KojUWrXmmJ/x55GMwvYZoI= |=tUpw |-END PGP SIGNATURE- |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call connected, cannot hear or speak - $20 for fix
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Supa wrote: Yes, it seems to fail when both extensions are external It seems more like a NAT problem than a script problem. Are the phones both connected via SIP? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoXZVS6d5vy0jeVcRAua7AJ9qOZuqlSEeqfWk5cG2v8nt7Buv5gCfVZSm to9odkqByRjDtwaKHogijEk= =g4bZ -END PGP SIGNATURE- ___ --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] Call connected, cannot hear or speak - $20 for fix
I using my provdier like so SIP/Telasip-gw4/5198843344 when bridging calls. All my local extensions work, so does disa and the like On 1/7/07, Matt Riddell (NZ) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1m usin Supa wrote: Yes, it seems to fail when both extensions are external It seems more like a NAT problem than a script problem. Are the phones both connected via SIP? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoXZVS6d5vy0jeVcRAua7AJ9qOZuqlSEeqfWk5cG2v8nt7Buv5gCfVZSm to9odkqByRjDtwaKHogijEk= =g4bZ -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hanging up a 3-way conference when middle user hangs up
Apparently asterisk's default way to a 3-way conference lets the user in the middle hangup and the other parties stay on the conversation. This is great some times but it creates quite a bit of problems when trunks dont have disconnect supervision or when trying to do accounting and billing on the user. Here is the 'normal' situation when a user tries the setup a 3 way conference once the user is already on the phone with someone he called. 1. Press 'Flash' on the phone. Party B will now be placed on hold and you will hear a dial tone. 2. Dial party C's number or a pre-configured speed dial followed by '#', (you can engage in conversation). 3. Press 'Flash' to join both C and B to a single conference. 4. When you place the phone's handset on-hook, party B and party C will remain in conversation. After step 4, B and C remain on the conversation and I am looking for a way to disable this without disabling 3 way calling. Basically I am looking for a way to force asterisk hang up both B and C once the 'middle' user hangs up so this will not leave channels stuck on trunks without disconnect supervision. Anyone know how to accomplish this? Any comments appreciated. ___ --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] Hanging up a 3-way conference when middle user hangs up
How does this compare to using the conference features on a SIP phone, say a Snom? I have used a Snom many times for an ad-hoc conference, without any troubles... PaulH On Sun, 2007-01-07 at 18:12 -0600, Lex Lethol wrote: Apparently asterisk's default way to a 3-way conference lets the user in the middle hangup and the other parties stay on the conversation. This is great some times but it creates quite a bit of problems when trunks dont have disconnect supervision or when trying to do accounting and billing on the user. Here is the 'normal' situation when a user tries the setup a 3 way conference once the user is already on the phone with someone he called. 1. Press 'Flash' on the phone. Party B will now be placed on hold and you will hear a dial tone. 2. Dial party C's number or a pre-configured speed dial followed by '#', (you can engage in conversation). 3. Press 'Flash' to join both C and B to a single conference. 4. When you place the phone's handset on-hook, party B and party C will remain in conversation. After step 4, B and C remain on the conversation and I am looking for a way to disable this without disabling 3 way calling. Basically I am looking for a way to force asterisk hang up both B and C once the 'middle' user hangs up so this will not leave channels stuck on trunks without disconnect supervision. Anyone know how to accomplish this? Any comments appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dimensioning a 50 sip phone installation
Erick Perez wrote: The customer found a VIA EPIA 1.2ghz with 1gb ram, one pci slot mini.itx. Can this equipment handle a sangoma/digium E1 card with 25 SIP ulaw phones +voicemail and *no* call recording? Make sure there're no interrupt sharing issues. My old EPIA 1GHz with 2 LAN and 6 USB, had no IO-APIC. It took lots of trial and error to make sure the digium card was not sharing an interrupt. Leo ___ --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] Re: asterisk-users Digest, Vol 30, Issue 7
On 1/3/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: -- Forwarded message -- From: Dan Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 3 Jan 2007 10:33:02 -0800 Subject: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released We've been holding back on this release to coincide with the Asterisk 1.4.0 release. This is mostly a compatibility release, but there are a few new features: * No longer requires register_globals in PHP * Separated code from configuration settings in ./lib/defines.php (hopefully this will make future upgrades easier) * Migrated all database interfaces to PEAR::DB which simplifies the code a bit and opens up the possibility of using other databases to host the scheduling DB (app_cbmysql is still only MySQL, but ODBC is planned/hoped for) * The conference monitoring code now uses the concise output from meetme list, improving the parsing of participant details. * Minor tweaks to improve the cbEnd.php script that enforces the conference duration, plays announcements and populates the conferencing CDRs. * Conference CDR records now store participant duration in seconds instead of a formatted string, allowing for further analysis (the web interface still formats the duration for display purposes) * App_cbmysql is updated to work with Asterisk 1.4.0 * App_cbmysql has it's own build environment now, no longer requiring a Makefile patch, etc... The new release can be found at: http://sourceforge.net/projects/web-meetme/ We do have a volunteer developer who will be maintaining the 2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and features that are not Asterisk version dependant will still be made available for older installations. Thanks, The Web-MeetMe development team... HI Dan. I have been using the Web Meet-Me 2.1.0 on Asterisk 1.2.12.1 for many months now and I am a big fan and I have been very happy with it. I want to try the v3.0.0 but I would like to know if there are specific steps I need to carry out to upgrade to the v3.0.0 on my current Asterisk 1.2.X? Warm Regards, Buki (Naija Man) ___ --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] Re: [Announce] Web-MeetMe 3.0.0 released
Sorry I forgot to change the subject line in my last posting! I have been using the Web Meet-Me 2.1.0 on Asterisk 1.2.12.1 for many months now and I am a big fan and I have been very happy with it. I want to try the v3.0.0 but I would like to know if there are specific steps I need to carry out to upgrade to the v3.0.0 on my current Asterisk 1.2.X? Warm Regards, Buki On 1/7/07, Naija Man [EMAIL PROTECTED] wrote: On 1/3/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: -- Forwarded message -- From: Dan Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 3 Jan 2007 10:33:02 -0800 Subject: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released We've been holding back on this release to coincide with the Asterisk 1.4.0 release. This is mostly a compatibility release, but there are a few new features: * No longer requires register_globals in PHP * Separated code from configuration settings in ./lib/defines.php (hopefully this will make future upgrades easier) * Migrated all database interfaces to PEAR::DB which simplifies the code a bit and opens up the possibility of using other databases to host the scheduling DB (app_cbmysql is still only MySQL, but ODBC is planned/hoped for) * The conference monitoring code now uses the concise output from meetme list, improving the parsing of participant details. * Minor tweaks to improve the cbEnd.php script that enforces the conference duration, plays announcements and populates the conferencing CDRs. * Conference CDR records now store participant duration in seconds instead of a formatted string, allowing for further analysis (the web interface still formats the duration for display purposes) * App_cbmysql is updated to work with Asterisk 1.4.0 * App_cbmysql has it's own build environment now, no longer requiring a Makefile patch, etc... The new release can be found at: http://sourceforge.net/projects/web-meetme/ We do have a volunteer developer who will be maintaining the 2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and features that are not Asterisk version dependant will still be made available for older installations. Thanks, The Web-MeetMe development team... HI Dan. I have been using the Web Meet-Me 2.1.0 on Asterisk 1.2.12.1 for many months now and I am a big fan and I have been very happy with it. I want to try the v3.0.0 but I would like to know if there are specific steps I need to carry out to upgrade to the v3.0.0 on my current Asterisk 1.2.X? Warm Regards, Buki (Naija Man) ___ --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] Re: Error on answer a SIP 401 message
Hi folks, Anyone already seen any problem like this ? Thanks. Fred Em Qua, 2007-01-03 às 16:24 -0300, Frederico Madeira escreveu: Hi, I'm a voip service provider and i'm setting up a asterisk box to register around 100 lines from my central softswitch. This asterisk box will be placed inside a customer and has a digium card to be interconected with customer's pabx. My problem is that when asterisk send register message, my softswitch return with sip 401 and asterisk should send a register message with Authorization in header. Only after 10, or 15 or 30 or 43 or 2 messages 401 asterisk start to send Authorization in header. This is a random time, don't follow any rule. This problem cause lines disregistration some times during a day. How can i solve this problem ? I use this parameters to register an account: register=number:[EMAIL PROTECTED]/number [fonar-number] type=friend context=default secret=pass username=number host=sip.provider.com fromuser=number fromdomain=sip.provider.com ;nat=yes ;insecure=very canreinvite=no ;qualify=1 dtmfmode=rfc2833 Thanks. signature.asc Description: Esta é uma parte de mensagem assinada digitalmente ___ --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] How to transfer Voicemail messages between 2 Asterisk servers
I have 2 simple asterisk servers linked over IAX. I want to be able transfer voicemail messages from my phone on Asterisk1 to another extension on the remote Asterisk2 by using the option 8 of the VoiceMail menu (transfer to another extension) ie: to transfer my voice mail messages in mailbox of SIP_PHONE1 on Atserisk1 to mailbox of SIP_PHONE2 on Asterisk2. SIP_PHONE1 --Asterisk1 ---IAX2-- Asterisk2 ---SIP_PHONE2 Asterisk1: v1.2.8 Asterisk2: v1.2.12.1 Any help will be appreciated. Thanks. Buki ___ --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] Hanging up a 3-way conference when middle user hangs up
As far as I know when I setup a 3-way on something like a cisco will disconnect everyone when the middle (person who setup the conference) hangs up. The problem I describe happens on ATAs and the like that uses flash to put on hold while setting up the second call. I am not sure about other phones other than cisco, polycom and a few others. Thanks! Ed On 1/7/07, Paul Hales [EMAIL PROTECTED] wrote: How does this compare to using the conference features on a SIP phone, say a Snom? I have used a Snom many times for an ad-hoc conference, without any troubles... PaulH On Sun, 2007-01-07 at 18:12 -0600, Lex Lethol wrote: Apparently asterisk's default way to a 3-way conference lets the user in the middle hangup and the other parties stay on the conversation. This is great some times but it creates quite a bit of problems when trunks dont have disconnect supervision or when trying to do accounting and billing on the user. Here is the 'normal' situation when a user tries the setup a 3 way conference once the user is already on the phone with someone he called. 1. Press 'Flash' on the phone. Party B will now be placed on hold and you will hear a dial tone. 2. Dial party C's number or a pre-configured speed dial followed by '#', (you can engage in conversation). 3. Press 'Flash' to join both C and B to a single conference. 4. When you place the phone's handset on-hook, party B and party C will remain in conversation. After step 4, B and C remain on the conversation and I am looking for a way to disable this without disabling 3 way calling. Basically I am looking for a way to force asterisk hang up both B and C once the 'middle' user hangs up so this will not leave channels stuck on trunks without disconnect supervision. Anyone know how to accomplish this? Any comments appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Some queries on g729 license.
Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? Regards, Liangliang ___ --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 360 auto answer
Hi,
I'm testing paging using snom 360.
Can someone correct my dialplan?
Regards,
Jason.
==
;exten = _99,1,SIPAddHeader(Call-Info:
Answer-After=0)
;exten = _99,n,SIPAddHeader(Call-Info:
sip:192.168.1.113\;answer-after=0)
;exten = _99,n,Dial(SIP/${EXTEN:2})
exten = _99,1,Set(__SIPADDHEADER=Call-Info:
answer-after=0)
exten =
_99,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = _99,n,Set(__ALERT_INFO=Ring Answer)
exten = _99,n,Dial(SIP/${EXTEN:2})
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RE: [asterisk-users] snom 360 auto answer
This is my code (that I copied form somewhere) for paging a group of
phones. By dialling 99 it will page phones 2101, 2102 and 2105.
Just include the context ext-paging in your dial plan and modify the
extension numbers and all should be good.
This works on Linksys Phones but should also work on Snoms.
I hope this helps you.
[ext-paging]
exten = PAGE2101,1,GotoIf($[ ${CALLERID(number)} = 2101 ]?skipself)
exten = PAGE2101,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)
exten = PAGE2101,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE2101,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2101,n,Dial(SIP/2101,5)
exten = PAGE2101,n(skipself),Noop(Not paging originator)
exten = PAGE2102,1,GotoIf($[ ${CALLERID(number)} = 2102 ]?skipself)
exten = PAGE2102,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)
exten = PAGE2102,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE2102,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2102,n,Dial(SIP/2102,5)
exten = PAGE2102,n(skipself),Noop(Not paging originator)
exten = PAGE2105,1,GotoIf($[ ${CALLERID(number)} = 2105 ]?skipself)
exten = PAGE2105,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)
exten = PAGE2105,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE2105,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2105,n,Dial(SIP/2105,5)
exten = PAGE2105,n(skipself),Noop(Not paging originator)
exten = Debug,1,Noop(dialstr is
LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]
aging)
exten =
99,1,Page(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/PAGE
[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Monday, 8 January 2007 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] snom 360 auto answer
Hi,
I'm testing paging using snom 360.
Can someone correct my dialplan?
Regards,
Jason.
==
;exten = _99,1,SIPAddHeader(Call-Info:
Answer-After=0)
;exten = _99,n,SIPAddHeader(Call-Info:
sip:192.168.1.113\;answer-after=0)
;exten = _99,n,Dial(SIP/${EXTEN:2})
exten = _99,1,Set(__SIPADDHEADER=Call-Info:
answer-after=0)
exten =
_99,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = _99,n,Set(__ALERT_INFO=Ring Answer)
exten = _99,n,Dial(SIP/${EXTEN:2})
__
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[asterisk-users] AMAFlags alwaysDocumentation (or 3 in astcdr mysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill)
Is anyone out there using AMAFlags? I'd like to set this field as a marker to distinguish different types of calls in CDRs, but can't seem to make it respond to the documented commands Set(CDR(amaflags)=bill) or SetAMAFlags(bill). I've googled this issue, seen others have had this problem with IAX, with different DB drivers for CDR records, etc. I'm using SIP and LOCAL channels, asterisk-1.4beta2 release (I don't think upgrading to current release will fix this problem, it's been around for years based on trouble reports), both text .csv and mysql astcdr.cdr types. Seems like a problem with basic AMAflags support in CDR. They always show up as DOCUMENTATION in the .csv text file, and they always show up as '3' in mysql. I hurt my brain trying to follow the layers of indirection in the source code for where this is actually set. With verbosity turned on in asterisk console I can see the SetAMAFlags function being run. Any tips, tricks, or pointers in the right direction? Thanks, Scott ___ --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] Problems with park
Hi. I've spent the past four hours on this... if it's a FAQ, I apologize. I am setting up a system with Asterisk 1.4, a TDM400 (3 FXO, 1 FXS) and 3 Aastra 480i CT phones. (1.4.1 firmware.) I have the system mostly working, but am still having trouble with a couple features. This email will deal with parking. First, I think I should point out that transferring calls seems to work fine. If I call into an FXO and have it ring on extension 102, I can answer the call and transfer it to extension 101. (Both Aastra 480i CTs) I have followed the (very brief) instructions on voip-info.org titled Asterisk Call parking. Basically, I confirmed that features.conf was already set up properly, and made sure parkedcalls was included in my local context. If I dial in via the FXO and answer the call on x102, then hit transfer 700, I hear the announcement 701. When I hang up the phone, the LCD displays Transfer Failed and I get a bunch of orphaned channels. I have made sure that all my Dial commands end with ,tT and have also added canreinvite=no to my sip.conf. Am I missing something simple? (probably) Can someone please point me in the right direction or let me know what config files you need to see? Thanks in advance. Oh, I just noticed, transferring _doesn't_ work fine... If I call from x101 to x102, and try to transfer the call to an extension set up with the Music on hold application, I get the same Transfer Failed message. Same thing if I call in via the FXO. Now I'm even more lost, but perhaps the problem isn't my parking config. ttyl srw ___ --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] AMAFlags alwaysDocumentation (or 3 in astcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill)
Hello Why not use the CDR(userfield) field instead. You can set that to any integer of your liking, and use that to identify the type of call. Jon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Keagy Sent: 8. januar 2007 06:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AMAFlags alwaysDocumentation (or 3 in astcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill) Is anyone out there using AMAFlags? I'd like to set this field as a marker to distinguish different types of calls in CDRs, but can't seem to make it respond to the documented commands Set(CDR(amaflags)=bill) or SetAMAFlags(bill). I've googled this issue, seen others have had this problem with IAX, with different DB drivers for CDR records, etc. I'm using SIP and LOCAL channels, asterisk-1.4beta2 release (I don't think upgrading to current release will fix this problem, it's been around for years based on trouble reports), both text .csv and mysql astcdr.cdr types. Seems like a problem with basic AMAflags support in CDR. They always show up as DOCUMENTATION in the .csv text file, and they always show up as '3' in mysql. I hurt my brain trying to follow the layers of indirection in the source code for where this is actually set. With verbosity turned on in asterisk console I can see the SetAMAFlags function being run. Any tips, tricks, or pointers in the right direction? Thanks, Scott ___ --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] Some queries on g729 license.
Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd ___ --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] Some queries on g729 license.
Hi, actutally it is kind of shareing storage, because we use drbd and vserver technology, the fail over is at vserver level, and vserver is synced through drbd storage. Regards, Liangliang Leo Ann Boon wrote: Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Interrupt rates and voip traffic
Hi, This is slightly off topic, but here I go any way... VoIP traffic has lot's of smaller packets, and since each packet can generate an interrupt, is there any way to determine the irq rates in a machine, and more importantly to know if I am hitting any of the limits in Linux or to determine how much interrupts per second can my box handle ? There seems to absolutely no information about his particular metric any where.. raj ___ --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] Some queries on g729 license.
Xue Liangliang wrote: Hi, actutally it is kind of shareing storage, because we use drbd and vserver technology, the fail over is at vserver level, and vserver is synced through drbd storage. drdb - that's what I suspected. Off the top of my head, the fastest way is to reactivate using the new master's MAC. The proper solution is to only use drdb for data that should be shared like the conf and database. The license key portion should not be on a device that's being mirrored by drdb. Leo ___ --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] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)
On 2007-01-07 01:23:22 -0800, Mark Coccimiglio [EMAIL PROTECTED] said: Marty, Where are you paying $1000 for a 1600 series Cisco? I can get you 20% off that price on any quantity (note: Sarcasam). Its not the 1990's anymore. You can get them on eBay ($50-150) for only slightly more then the Linksys. The performance is rock solid. Three-quarters of the world have used them for decades. I know of units running 2 and 3 YEARS between reboots. The power company reboots my equipment more then I do. Ok it is true that Cisco does not support the models anymore, but you can't buy a services contract for a linksys router either. It can sometimes be a little difficult to configure without any technical knowledge but that is what most of us get paid for. It does impress the customer when you bring in the grey box labled Cisco. As for performance just try to put 50 people behind a linksys/netgear/dlink. I've used 1605R supporting +100 users. Not even a blink. Finally, untill everyone is using 10Mps FTTH the broad band link is still the slowest part of the connection. Not to shabby for antiquated technology. Ok, Thanks for the pricing update. I am decidedly old school and sometimes get stuck with old information. I did see people selling these for $800. But I guess the web is full of antiquated pricing info also. Thanks for the idea. Marty PS Any ideas on inexpensive wireless APs? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with park
I have followed the (very brief) instructions on voip-info.org titled Asterisk Call parking. Basically, I confirmed that features.conf was already set up properly, and made sure parkedcalls was included in my local context. If I dial in via the FXO and answer the call on x102, then hit transfer 700, I hear the announcement 701. When I hang up the phone, the LCD displays Transfer Failed and I get a bunch of orphaned channels. Are you doing a blind transfer or attended transfer? I'm assuming you're using the phone's transfer button. You may need to press transfer a second time to complete the transfer. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bypass menu for certain numbers?
Just want to comment that this worked great.
Thanks!
I even added in a little cepstral thing so it says Hello Caller
Name, Please wait while we bypass the menu and dial directly to make
them feel special :)
Matt G
On 04/01/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Donnerstag, den 04.01.2007, 17:47 -0500 schrieb Noah Miller:
Let's say you use the Asterisk DB() stuff for your caller ids, storing
them in a branch called book like
book/16175551234 John Doe
book/12125559876 Jane Miller
Then you could go with a logic like
exten = s,1,Set(CALLERID(name)=${DB(book/${CALLERID(num)})})
exten = s,2,GotoIf($[${CALLERID(name)} = ]?3:100)
exten = s,3,---your business customers go here---
exten = s,100,--- friends go here ---
I don't think this would do what you want it to do. I think you
should leave the caller id name out of it, and just use numbers.
Also, I don't think you really want to change the CALLERID of the
channel at all, just use it to compare.I like Tzafrir's
ex-girlfriend method.
The positive thing about changing CALLERID(name) is that in case a name
is present it can be displayed on the phone.
Just curious: Is there any reason to *not* use CALLERID(name)?
Of course, you could compres lines 1 and 2 to something like
exten = s,1,GotoIf($[${DB(book/${CALLERID(num)})} = ]?3:100)
I would prefer storing the name, once retrieved from the database, to
later hand it over to the phone - so why not just putting it into the
appropriate variable?
I'd like to have it only
be one goto for direct dial, and one for the main menu instead of
having to manually input all those numbers, and add new ones when
required.
Whether you enter them into the dialplan with the ex-girlfriend
method, put them in a DB, or put them in a goto statement, you're
going to have to add the new numbers at least once. I think it's the
same amount of work to add new numbers using any method. The
ex-girlfriend method seems a little more in tune with the 1.4 style
(non-jumping) dialplans.
One difference is at which time the actualisation takes place. As I
understand, changes in the database could be easily done through a
web-interface, script or whatever without any complex dialplan file
rewriting logic, with changes taking place immediately. Changes to the
dialplan need an extensions reload...
Besides, from an administrative point of view, putting a list into a
database / table probably makes the dialplan easier to read than having
dozens of lines only differing in the caller id part.
My asterisk dialplan for all my phone communication needs heavily relies
on database functions. Any callrouting (incoming SIP calls route to
which internal ids), call forwarding, voicemail delay/activation, call
groups, automated reminder calls - all that can be easily changed by
ssh'ing into the asterisk box, doing asterisk -rx database set
something, voila.
Your mileage is certainly a different one, and I thankfully take ideas
for improvement.
Best regards,
Anselm
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Re: [asterisk-users] Some queries on g729 license.
Hi, leo, I will try the following solution that seperate /usr/lib/asterisk/modules in another patition other than drbd, then register the licenses on both server. not sure where the license key acutally lies in? Regards, Liangliang Leo Ann Boon wrote: Xue Liangliang wrote: Hi, actutally it is kind of shareing storage, because we use drbd and vserver technology, the fail over is at vserver level, and vserver is synced through drbd storage. drdb - that's what I suspected. Off the top of my head, the fastest way is to reactivate using the new master's MAC. The proper solution is to only use drdb for data that should be shared like the conf and database. The license key portion should not be on a device that's being mirrored by drdb. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] jitterbuffer on sip.conf
In iax.conf there is option jitterbuffer how about sip protocol ? Are jitterbuffer can configure in sip.conf ? Thanks, for your share ___ --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] jitterbuffer on sip.conf
[EMAIL PROTECTED] wrote: In iax.conf there is option jitterbuffer how about sip protocol ? Are jitterbuffer can configure in sip.conf ? Thanks, for your share If you upgrade to 1.4, there is a jitterbuffer available now for the SIP channel. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
