[asterisk-users] Test asterisk from behind my firewall
I have an asterisk server at home. I'd like to test one just installed elsewhere. Both servers are behind firewalls. I can see the session start in CLI, my congratulations is apparently playing and RTP is being sent. Hearing no audio. Can send key presses and see audio playing changed. Peer audio RTP is at port 198.145.28.177:10180, but that never shows at the client side, behind a linksys wrt54g, ver 1. w/ latest firmware update. My belief is this should be possible, as the SIP phone is registered to my asterisk box inside my home network, asterisk should stay in the middle and forward the RTP packets to my laptop... am I totally off base? If so, what are some key elements to make that happen? I'll stop now, before I get ignored for being too verbose. '-) Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Test asterisk from behind my firewall [SOLVED]
On Mon, 16 Mar 2009 23:00:32 -0700 Michael Higgins li...@evolone.org wrote: I have an asterisk server at home. I'd like to test one just installed elsewhere. And did succeed just after emailing, of course. :( Sorry for the noise! -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with large user base?
Vincent Li wrote: Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. Asterisk can scale to 10.000 users. Its probably about the maximum you could do on a quite powerful server if you don't need TDM hardware, but better would be to use a cluster, the database used would then eventually become the limit to the scaling. I have no experience with SipX so i can't say if it will scale better without clustering. The project manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running in my office for small user base, I don't have experience with large scale Asterisk implementation. I know little about sipX. Does anyone in the community has any input about this? Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for a patch cable for my SPA941 Phones
Hi all, i know this question is not directly asterisk related - but i have no idea where else to ask. We do have around 50 pieces of LinkSys SPA941 - these phones do have a 2.5mm plug connection - and we do have many many headsets we used with normal PC's before (so 2x3.5mm plug connection). Does anyone here know where i can get an adapter 1x2.5mm - 2x3.5mm ? Or can anyone here tell me where to get good (and not to expensive) 2.5mm plug connection binaural headsets ? best regards, Wolfgang ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] url in dial command: how does it work?
Hello Giorgio, you simply pass that parameter along so that from the QueueMetrics agent page you get that URL opened automagically when you get a call. It's for interfacing to external CRM apps, usually passing the agent code that handles the call, the Asterisk unique-id and the caller-id for database matching. A lot of people using both the free and the commercial versions of QM use it and it seems to be working just fine :) the biggest problem we have with this is that this is not supported by FreePBX GUI and its derivatives (but we are going to make the configuratyion at the queue level optional, while still retaining the functionality). I hope this helps, l. 2009/3/16 Giorgio Incantalupo gincantal...@fgasoftware.com Hi Tim, I've made a test with 2 Asterisks and the 2 consoles showed me an HTML packet sent and one received. This does not work with the SIP protocol. The idea was to understand what was it for (I suppose someone did it for some purpose...), then how to use it to improve our solution (es: open pop ups) but we use SIP phones which do not support that URL parameter. I know queuemetrics use it but I cannot undestand how since tha URL parameter is passed to the called party while queuemetrics reads the queues.log file. BTW thanks for your time. Giorgio Tim Panton wrote: Oh sorry, I wasn't clear. The IAX protocol has a frame type for sending this URL info. Skype has an attribute for it. The intention is (I think) to be able to forward the URL for the customer (in the corporate CRM system) to the agent answering a call on a softphone. Some of the IAX softphones support this. What were you planning to do with it. Tim. On 16 Mar 2009, at 13:04, Giorgio Incantalupo wrote: Hi Tim, ok, but I think the big question is...what is the URL for? It seems I need a special device...but which? What kind of device do you use? Thanks. Giorgio Tim Panton wrote: Use IAX :-) In principle chan_skype could also support it. T. On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: Hi, Does anybody knows where I can find some docs about how to make the URL parameter inside the Dial command work? I tried to make some tests with a sip phone without success: the sip debug shows no URL inside sip packets. :( Any hint appreciated. :) Thank you Giorgio ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Plastic Water Bottles
Hi, Sorry for following on this off-topic, but, On Mon, Mar 16, 2009 at 08:49:53PM -0600, drew einhorn wrote: The plastics industry says polycarbonate bottles are safe. http://www.bisphenol-a.org/about/faq.html#g I'm sure Maggie and here friends would say ALL plastic bottles are very dangerous. This lady seems to be at a reasonable middle ground. http://trusted.md/blog/vreni_gurd/2007/03/29/plastic_water_bottles http://www.cancer.ca/Canada-wide/About cancer/Cancer myths/Reusing disposable water bottles.aspx?sc_lang=en is also relevant. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
On Mon, 16 Mar 2009, Gavin Henry wrote: Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue lines should this go down. Gavin, You won't get 12 concurent G711 calls over a standard ADSL line in the UK. If you're on an ADSL2+ service you may get up to 1.1Mb/sec upload speed, but even then, 12 * 80 = 960Kb/sec which is really pushing it, so use G729, or get that 2Mb SDSL line in. Make sure it's a decent ISP too. Using IAX will give you a few extra channels though as the IP overhead is less. What is key is billing information and the ability for a receptionist to see all active calls and do transfers etc. Much like the Flash Operator Panel. Desktop Software may also be needed for this purpose or can be done via a traditional bank of lines on an IP phone accessory module. Have a look at: http://www.astassistant.com/ rather than FOP. Even has a Linux client which is nice... If anyone has any ideas on the best way to put this together, I'm all ears ;-) The consultant in me says Pay someone to do it for you :) However it's not that hard to do and setup if youve done something similar in the past - and your budget is tight. If you know you're going to get more of these, then go for it - spend your time on the software and front-end for the the first one, then the rest are clones... I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra 53i phones. There's a £4k budget for this (still waiting for more into)which will include the networking connection and equipment. If I can afford it I normally go Sangoma with Echo cancellation, but as it's a fallback service, so I'm not bothered. When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Cheers, Gordon -- www.drogon.net___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with large user base?
17 mar 2009 kl. 07.26 skrev zoach...@securax.org: Vincent Li wrote: Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. Asterisk can scale to 10.000 users. Its probably about the maximum you could do on a quite powerful server if you don't need TDM hardware, but better would be to use a cluster, the database used would then eventually become the limit to the scaling. I have no experience with SipX so i can't say if it will scale better without clustering. We've built several solutions for carriers and universities that scales to a very large userbase. It's certainly possible, but needs good design and preparation, but that applies to all projects with that many users, regardless if it's open source. /O --- http://edvina.net - Asterisk/OpenSER/Kamailio consulting and training Sollentuna, Sweden ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bristuff bug or feature ? (Was: Are .call files working with extensions.ael ? bristuff problem)
On Tue, Mar 17, 2009 at 12:28:25AM +0100, Olivier wrote: Hi, Is the following behaviour a bug or a feature ? A bug. Those two fields should be optional. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mobile centrex solution
anyone know of a solution where mobile handsets out roaming the pstn cellular network can be used and treated as full fleged centrex extentions, i.e. I can transfer a call that comes in on a wired centrex copper pair out to a cell phone and the cell phone can transfer the call back or vice versa where the cell phone recieves the call directly and can transfer to the office all without hairpinning the call? essentially when the call is transfered I'd like to have asterisk get out of the call path but still have the capability to transfer the call back to asterisk and it's attached office phones. Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with large user base?
2009/3/17 zoach...@securax.org zoach...@securax.org Vincent Li wrote: Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. Asterisk can scale to 10.000 users. Its probably about the maximum you could do on a quite powerful server if you don't need TDM hardware, but better would be to use a cluster, the database used would then eventually become the limit to the scaling. I have no experience with SipX so i can't say if it will scale better without clustering. The project manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running in my office for small user base, I don't have experience with large scale Asterisk implementation. I know little about sipX. Does anyone in the community has any input about this? Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello i suggest opensips/kamalio for register server role and asterisk for a voicemail server and to pstn/pri/whatever gateway. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
Hi all, maybe I find the problem and the solution. I move the following parameters on section [general]: [general] port=5060 bindaddr=0.0.0.0 context=default language=it limitonpeers=yes notifyringing=yes and then on SIP account I put this: [intphones](!) type=friend qualify=yes host=dynamic callgroup=0 pickupgroup=0 dtmfmode=info [10](intphones) context=office username=10 secret=1234 subscribecontext=BLF_group call-limit=1 and this works! When someone call SIP/10, and then I call again SIP/10, I find it busy. On the other side, when SIP/10 make a call, and then I call again SIP/10, I find it busy. And that's ok! But there is another little problem. On Aastra phone (on other phones I don't try yet), the xfer button doesn't work, until I set call-limit=2, but making this I find the phone not busy. Anyone know how to use busy-level parameter or some other useful parameters? Thanks all Marco 2009/3/16 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Mon, 16 Mar 2009, Olivier wrote: 2009/3/16 Gordon Henderson gordon+aster...@drogon.net gordon%2baster...@drogon.net gordon%2baster...@drogon.net gordon%252baster...@drogon.net On Mon, 16 Mar 2009, Marco Sambo wrote: Hi, I have a question. How can I configure my sip.conf to make a SIP phone busy on incoming and outcoming calls? I explain my problem. When SIP phone receive a call and then I try to call that phone, I find it busy. When SIP phone make a call and I try to call that phone, I find it avaible and it rings but I want to find it busy. Disable call-waiting inside the phone. Doesn't call-limit=1 force the same behaviour ? It appears to limmit the number of outgoing calls from that phone and independantly the number of inoming calls. So a phone can make an outgoing call, and still take an incoming call, and vice-versa, with call-limit=1 I also found early versions of this buggy in that it didn't seem to properly decrement the counter on hang-up, so is call-limit was set to 3, then that phone could only take 3 calls, one after the other, before it would be premenantly busyd, but this was a long time back, and it might have been something I was foing, but since then I always turned call-waiting off on the phones when users didn't want multiple call features. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to get 60+ analogue extensions.
2009/3/16 Alex Balashov abalas...@evaristesys.com I don't know how good Asterisk's GR.303 support, but you could use DLCs as well. However, that's a lot of complexity and (seemingly) immature functionality liability to achieve the same end you'd get with a channel bank. The only benefit is that DLCs are specifically for oversubscription, whereas on PRIs you'd be doing one timeslot per one POTS line on the trunk side. On Mon, 16 Mar 2009 18:48:10 -0400, C F shma...@gmail.com wrote: Channel Banks would be the way I would do it. On Sun, Mar 15, 2009 at 3:12 AM, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. I can see the Digium AEX2400 with 24 lines, physically they are all very deep, if I had 3 of these in a server it would seem straight forward assuming the motherboard doesn't haven't anything get in the way Equally the Digium TDM2400P supports 24 lines and physically requires similar space The Sangoma A400 provides 24 ports but uses two slots, having 3 of these in a server looks like I need to pick the server carefully. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. I am just interested in any recommendations for server hardware and card combinations that are currently in use. Also if anyone has provided call data out to the RMS system ( http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to hear how it worked. Thanks very much Cheers Duncan ___ xorcom 2x32 fxs and done ;) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with large user base?
How are SipX solutions sold to Universities ? Are those solutions directly sold by the company mostly contributing to SipX development, by licenced partners or by local integrator, not having much commercial link with SipX editor ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
We can put about 9/10 calls using SIP/gsm through our BT Business Network ADSL package connection (832kbit upstream, £65/month) before you notice the quality starting to drop, but you could always get two connections and bond them together into one using openvpn or some other method if you wanted to. 2009/3/17 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Mon, 16 Mar 2009, Gavin Henry wrote: Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue lines should this go down. Gavin, You won't get 12 concurent G711 calls over a standard ADSL line in the UK. If you're on an ADSL2+ service you may get up to 1.1Mb/sec upload speed, but even then, 12 * 80 = 960Kb/sec which is really pushing it, so use G729, or get that 2Mb SDSL line in. Make sure it's a decent ISP too. Using IAX will give you a few extra channels though as the IP overhead is less. What is key is billing information and the ability for a receptionist to see all active calls and do transfers etc. Much like the Flash Operator Panel. Desktop Software may also be needed for this purpose or can be done via a traditional bank of lines on an IP phone accessory module. Have a look at: http://www.astassistant.com/ rather than FOP. Even has a Linux client which is nice... If anyone has any ideas on the best way to put this together, I'm all ears ;-) The consultant in me says Pay someone to do it for you :) However it's not that hard to do and setup if youve done something similar in the past - and your budget is tight. If you know you're going to get more of these, then go for it - spend your time on the software and front-end for the the first one, then the rest are clones... I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra 53i phones. There's a £4k budget for this (still waiting for more into)which will include the networking connection and equipment. If I can afford it I normally go Sangoma with Echo cancellation, but as it's a fallback service, so I'm not bothered. When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Cheers, Gordon -- www.drogon.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Olivier wrote: Hi, I've been playing with T.38. I observed that mostly but not always, it's the calling endpoint that reINVITE the other party to drop current SIP/G711 session and start a new T.38. But sometimes, it's also the callee party that reINVITE the calling party. Which is the standardized or most common, way to start a T.38 session ? Shall it come from callee or from caller ? Regards Fax transmission can be initiated from any one of the parties. AFAIK T.38 as well as the PSTN fax standards do not show any preference whether fax transmission is requested from a or b party. In practice, the caller usually initiates a fax transmission, but this doesn't mean that the called party cannot initiate it, too. Best regards, Vlasis Hatzistavrou. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk now and switchvox
What is the status of asterisk now and switchvox now that digium owns both? Is it expected that both will stay in continued development for the long term? why would someone use one over the other? From what I've seen both seem easier to use than trixbox/freepbx which I found so confusing as to go back to streight dialplan scripting in bare asterisk. (seemed like a non technical person would have a chance at admining asterisk now or switchvox where I would never give a trixbox/freepbx system to a non technical user to admin) Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA react to phone but unresponsive to fax modem [SOLVED]
2009/3/17 Olivier oza-4...@myamail.com 2009/3/16 Olivier oza-4...@myamail.com Hi, I'm rather new to this domain so I may be doing stupid things without being concious of that. I've got a Patton MATA I'm trying to setup as T.38 fax adapter. Whenever I connect a fax machine (Dell MFP1815dn) or a phone to it, I can successfully send a fax or talk to the other end. Whenever I connect a fax modem (Dell Inspiron 6400 laptop), I keep getting No signal. Line is busy or disconnect from Windows XP fax application. Whatching SIP trafic from this Patton MATA, I can see no single SIP is leaving the box so I'm certain issue relates to analog line settings but I'm mostly lost with things like Ring Polarity, Ring settings and so on. I tried to mimic settings from an SPA3102 with which I can either fax from fax machine or fax application but I'm unsuccessful at the moment. 1. Can you explain what is going on ? 2 What would you say reading this : Ring waveform: trapezoid Ring frequency: 20 Ring voltage: 85 Reducing voltage to 60 made the fax-modem reply. This is a bit strange as this value is quite different from the one working with SPA3102. FXS input gain: -6 FXS output gain: -6 (I copied those values from SPA3102 into MATA) Best regards Changing FXS input gain and FXS output gain from -6 to -12 improved things as I could fax out in T.38 with both ATAs and fax endpoints ! But for incoming faxes, modem connected to M-ATA remains silent and idle whenever the M-ATA receives a fax call : I can see incoming SIP signal arriving into the ATA but it seems no analog signal is going out from it. (using SPA3102, faxes are correctly received). How is called the signal an ATA uses when it wants to wake an analog phone or a fax machine up ? Is it correct to think the same electrical signal is sent whatever the analog endpoint is ? What could explain a phone is ringing at one and a fax modem remains idle ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird issue with outbound calls and MOH
Hi, We have a PRI Trunk (physical E1) and we are getting some rather weird and very isolocated problems. On outbound calls to specific numbers, it would seem to me that DTMF from the remote side is affecting the local asterisk system. Basically what happens: - We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System - Remote Answers, and converse - Remote sends DTMF on their site to transfer call - Our * system initiates on hold with our on hold music - ZAP channel drops, followed shortly after by the SIP channel. Zaptel configs are attached too. A trace of a call where this happened is below (DTMF debug logging is also enabled, and yet there is no indication of a DTMF being received): [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [0117709...@from-internal:1] Macro(SIP/8647-b6f96650, user-callerid|SKIPTTL|) in new stack [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-user-callerid:1] Set(SIP/8647-b6f96650, AMPUSER=8647) in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: Set [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-user-callerid:2] GotoIf(SIP/8647-b6f96650, 0?report) in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: GotoIf [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-user-callerid:3] ExecIf(SIP/8647-b6f96650, 1|Set|REALCALLERIDNUM=8647) in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: ExecIf [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-user-callerid:4] Set(SIP/8647-b6f96650, AMPUSER=8647) in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: Set [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-user-callerid:5] Set(SIP/8647-b6f96650, AMPUSERCIDNAME=Ntombi Njongo) in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: Set [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-user-callerid:6] GotoIf(SIP/8647-b6f96650, 0?report) in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: GotoIf [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-user-callerid:7] Set(SIP/8647-b6f96650, AMPUSERCID=8647) in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: Set [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-user-callerid:8] Set(SIP/8647-b6f96650, CALLERID(all)=Ntombi Njongo 8647) in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: Set [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-user-callerid:9] Set(SIP/8647-b6f96650, REALCALLERIDNUM=8647) in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: Set [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-user-callerid:10] GotoIf(SIP/8647-b6f96650, 1?continue) in new stack [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Goto (macro-user-callerid,s,19) [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: GotoIf [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-user-callerid:19] NoOp(SIP/8647-b6f96650, Using CallerID Ntombi Njongo 8647) in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: Noop [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [0117709...@from-internal:2] Set(SIP/8647-b6f96650, _NODEST=) in new stack [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [0117709...@from-internal:3] Macro(SIP/8647-b6f96650, record-enable|8647|OUT|) in new stack [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-record-enable:1] GotoIf(SIP/8647-b6f96650, 1?check) in new stack [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Goto (macro-record-enable,s,4) [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: GotoIf [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-record-enable:4] AGI(SIP/8647-b6f96650, recordingcheck|20090313-133037|1236943837.1282) in new stack [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [Mar 13 13:30:37] VERBOSE[28294] logger.c: recordingcheck|20090313-133037|1236943837.1282: Outbound recording not enabled [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- AGI Script recordingcheck completed, returning 0 [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: AGI [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-record-enable:5] MacroExit(SIP/8647-b6f96650, ) in new stack [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [0117709...@from-internal:4] Macro(SIP/8647-b6f96650, dialout-trunk|3|0117709800||) in new stack [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [...@macro-dialout-trunk:1] Set(SIP/8647-b6f96650, DIAL_TRUNK=3) in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: Set [Mar 13 13:30:37] DEBUG[28294] func_db.c: DB: AMPUSER/8647/pinless not found in database. [Mar
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
On Tue, 17 Mar 2009, Geraint Lee wrote: We can put about 9/10 calls using SIP/gsm through our BT Business Network ADSL package connection (832kbit upstream, £65/month) before you notice the quality starting to drop, but you could always get two connections and bond them together into one using openvpn or some other method if you wanted to. Ugh. GSM )-: I've never really had much luck with BT as an Internet provider either - their wholesale network - good, retail broadband, bad... In theory, you should be able to get 10 G711 SIP calls over a business quality 830Kb/sec upload ADSL line. I get 9 on my test setup before any packet loss. I managed 11 calls using IAX over the same line before loss. (Entanet ADSL and a Draytek router - £25 a month) Intersting idea re. using openvpn or similar.. I have sites with 3 ADSL connections - one for incoming calls, one for outgoing and one for general office use.. That works when the call numbers in/out is relatively balanced though. I know of a local company who're regularly putting 20 concurrent calls over the same broadband setup using G729... Gordon 2009/3/17 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Mon, 16 Mar 2009, Gavin Henry wrote: Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue lines should this go down. Gavin, You won't get 12 concurent G711 calls over a standard ADSL line in the UK. If you're on an ADSL2+ service you may get up to 1.1Mb/sec upload speed, but even then, 12 * 80 = 960Kb/sec which is really pushing it, so use G729, or get that 2Mb SDSL line in. Make sure it's a decent ISP too. Using IAX will give you a few extra channels though as the IP overhead is less. What is key is billing information and the ability for a receptionist to see all active calls and do transfers etc. Much like the Flash Operator Panel. Desktop Software may also be needed for this purpose or can be done via a traditional bank of lines on an IP phone accessory module. Have a look at: http://www.astassistant.com/ rather than FOP. Even has a Linux client which is nice... If anyone has any ideas on the best way to put this together, I'm all ears ;-) The consultant in me says Pay someone to do it for you :) However it's not that hard to do and setup if youve done something similar in the past - and your budget is tight. If you know you're going to get more of these, then go for it - spend your time on the software and front-end for the the first one, then the rest are clones... I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra 53i phones. There's a £4k budget for this (still waiting for more into)which will include the networking connection and equipment. If I can afford it I normally go Sangoma with Echo cancellation, but as it's a fallback service, so I'm not bothered. When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Cheers, Gordon -- www.drogon.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Direct Dial-Out and CDR destination numbers
Hi, as German phone numbers are variable_length, I need to use direct dial-out. The problem is that only the part which appears in extensions.ael (and thus in the argument to Dial()) is logged to the call data record. What I want, obviously, is for the Dial() app to append the additional digits to the CDR's destination number. Is that possible? -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and ericsson e1 connection how to??
Thanks for the reply.. You should be able to get support from the people who sold you the card. You need to configure 2 files (I'm looking at an old system, so they have the zaptel style names). My files are below - the thing to note is the span 1,1,0, the second 1 tells you that the span is a timing source, externally clocked. Whatever i do in timing source parameter, it still shows as internal clock on dahdi_tool. Depending on the mode that your Ericsson is in, you may need to change signalling=pri_cpe to signalling=pri_net As i see other devices working with ericsson (gsm router) configured as TE (cpe) so i configured it like this too. And then created an incoming route to one of the extensions from this dahdi channel. But i got busy signal when i try to dial from ericsson side... /etc/asterisk/zapata.conf: ; Configuration file [channels] ; ; Default language ; language=en context=ntl switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes group=1 callgroup=1 pickupgroup=1 ;echocancel=256 ;channel = 1-6 channel = 1-15,17-31 and /etc/zaptel.conf : span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone = uk On 16 Mar 2009, at 18:11, Oguzhan Kayhan wrote: Hello, I am trying to install my E1 card to make a conection with an Ericsson MD-110 PBX. I installed dahdi drivers as: dahdi_hardware pci::04:08.0 wcte12xp-d161:8000 Wildcard TE121 ran dahdi_genconf and it created all my e1 ports. On the other side i also configured the pbx to communicate with TE121. On ericsson side, i have no error messages. On asterisk side, no error messages. But when i try to create a dahdi trunk, and dial it from asterisk , no call can be made. and also, when i try to call from ericsson side, i get line busy message as soon as i dial the number. Is there any guide that can help me in installing that card? PS: Whatever i made in SPAN config, everytime the only thing i see was Internal clock on dahdi_tool . How can i make my e1 card master (or slave whatever) instead of internal clock?? and other thing i wonder, if i create a span like span=1,0,0,ccs,hdb3 is it zap/g1 in zaptel(dahdi) conf menu in asteriskgui???(or freepbx) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Mon, 16 Mar 2009, Gavin Henry wrote: Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue lines should this go down. Gavin, You won't get 12 concurent G711 calls over a standard ADSL line in the UK. If you're on an ADSL2+ service you may get up to 1.1Mb/sec upload speed, but even then, 12 * 80 = 960Kb/sec which is really pushing it, so use G729, or get that 2Mb SDSL line in. Make sure it's a decent ISP too. Using IAX will give you a few extra channels though as the IP overhead is less. Thanks. We're waiting to hear abou twhat we can provide. We use Gradwell for termination and their ADSL. DSL Premium M does 2.5 up, but I'll limit this to 10 calls to be safe. What is key is billing information and the ability for a receptionist to see all active calls and do transfers etc. Much like the Flash Operator Panel. Desktop Software may also be needed for this purpose or can be done via a traditional bank of lines on an IP phone accessory module. Have a look at: http://www.astassistant.com/ rather than FOP. Even has a Linux client which is nice... Looks good. Just tested it on VirtualBox for box. If anyone has any ideas on the best way to put this together, I'm all ears ;-) The consultant in me says Pay someone to do it for you :) However it's not that hard to do and setup if youve done something similar in the past - and your budget is tight. If you know you're going to get more of these, then go for it - spend your time on the software and front-end for the the first one, then the rest are clones... Yeah. I normal use PBXinAFlash for this. Just the receptionist part that was missing and maybe add on a2billing. I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra 53i phones. There's a £4k budget for this (still waiting for more into)which will include the networking connection and equipment. If I can afford it I normally go Sangoma with Echo cancellation, but as it's a fallback service, so I'm not bothered. When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. Yeah, don't really like them though. I could go down to a 51i for £67 ex VAT. You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. What would that come in at? A Dell T100 is £300 ex VAT for 160GB, 1GB RAM and a Dual Core Intel® Pentium® E2220; 2.4GHz with 3yrs nxt bday. A 4 port FXO card is £126.95 ex vat. Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! Thanks. I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Yeah, as I planned, but not for this project. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Tue, 17 Mar 2009, Geraint Lee wrote: We can put about 9/10 calls using SIP/gsm through our BT Business Network ADSL package connection (832kbit upstream, £65/month) before you notice the quality starting to drop, but you could always get two connections and bond them together into one using openvpn or some other method if you wanted to. Ugh. GSM )-: I've never really had much luck with BT as an Internet provider either - their wholesale network - good, retail broadband, bad... In theory, you should be able to get 10 G711 SIP calls over a business quality 830Kb/sec upload ADSL line. I get 9 on my test setup before any packet loss. I managed 11 calls using IAX over the same line before loss. (Entanet ADSL and a Draytek router - £25 a month) Intersting idea re. using openvpn or similar.. I have sites with 3 ADSL connections - one for incoming calls, one for outgoing and one for general office use.. That works when the call numbers in/out is relatively balanced though. I know of a local company who're regularly putting 20 concurrent calls over the same broadband setup using G729... Yeah, we use g.729 ourselves too. Gavin. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
I use Polycom 320s. They have PoE, 2 lines, great sound quality and they work very well with Asterisk. They are also about $85 each. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Monday, March 16, 2009 6:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Good phone near $125 I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
On Tue, 17 Mar 2009, Gavin Henry wrote: 2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Mon, 16 Mar 2009, Gavin Henry wrote: When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. Yeah, don't really like them though. I could go down to a 51i for £67 ex VAT. Grandstreams aren't to everyones liking, this is true... You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. What would that come in at? A Dell T100 is £300 ex VAT for 160GB, 1GB RAM and a Dual Core Intel® Pentium® E2220; 2.4GHz with 3yrs nxt bday. Under £200 from someone like http://linitx.com/ I don't put disk drives in my boxes though - they boot out of flash. I guess with the Dell, you have on-site or next day replacement if you take that deal though. A 4 port FXO card is £126.95 ex vat. (From voipon by the looks of that price ;-) Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! Thanks. I've been approcached by a client who wants a sort of hotel billing system though - tailored to their needs - it's for a retirement home sort of thing. I suggested they just did a fixed-price deal with the inmates, but that didn't go down well. They want to account for everything to the last penny )-: I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Yeah, as I planned, but not for this project. Good luck! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Vlasis Hatzistavrou (KTI) wrote: Olivier wrote: Hi, I've been playing with T.38. I observed that mostly but not always, it's the calling endpoint that reINVITE the other party to drop current SIP/G711 session and start a new T.38. But sometimes, it's also the callee party that reINVITE the calling party. Which is the standardized or most common, way to start a T.38 session ? Shall it come from callee or from caller ? Regards Fax transmission can be initiated from any one of the parties. AFAIK T.38 as well as the PSTN fax standards do not show any preference whether fax transmission is requested from a or b party. In practice, the caller usually initiates a fax transmission, but this doesn't mean that the called party cannot initiate it, too. Best regards, Vlasis Hatzistavrou. Hey, why bother looking at a spec when its so much more fun to make it up as we go along? T.38 says that if the call starts in audio mode it is the called end which should initiate a re-invite to change from audio to T.38. This makes sense, as that is the end which has the best chance of figuring out if a FAX machine answers the call. In practice many T.38 implementations will send out a re-invite when they are the calling side, so any practical implementation has to allow for this. Clashes are possible, if both ends send re-invite, and this is not always handled properly Also many implementations will only listen for a FAX machine at the beginning of a call, so if a human answers and later presses the start button on their FAX machine the T.38 gateway might miss this. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
On Tue, 17 Mar 2009, Gavin Henry wrote: 2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Tue, 17 Mar 2009, Geraint Lee wrote: I know of a local company who're regularly putting 20 concurrent calls over the same broadband setup using G729... Yeah, we use g.729 ourselves too. The issues I've had have been when theres transcoding going on that you can't control - ie. outside your network, so I can go point to point from end-user phone to the people I peer with, but if they then transcode to G711 to go to the PSTN, it's OK, but if it then gets transcoded to GSM for a mobile, or back to G729 to go to an expensive overseas location, then quality does suffer )-: Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXP2000 BLF
We have a system running SNOM 360s, and BLF works fine. We are trying Grandstream GXP2000s and like the phones for what they are, but can't get the BLF to work. The IB just says to set to BLF and put in the phone number. We have tried variations like adding @xxx.xxx.yyy.zzz, but no lights light. Does anyone have the magic incantation to get the BLF to work? Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF troubles
I've been using one of the popular asterisk ISO distributions for a couple of years and DTMF had always worked. I recently switched to another asterisk ISO distribution, and outbound DTMF is no longer working. After doing a bit of digging, I noticed that the new distribution wasn't setting any sort of dtmfmode at all, anywhere. The old distribution had rfc2833 set in sip.conf for the phone extensions, so I thought I'd set the dtmfmode and see if that helped. Unfortunately, it did not. I then went ahead and set dtmfmode=inband in sip.conf and iax.conf for the phones and trunk to provider, respectively. At that point, I was finally able to hear DTMF tones when I called out to my cell from my 7960 but they still weren't enough to trigger the menu of an IVR outside of the * system. Jitterbuffer is disabled on the IAX trunk to the provider. DTMF seems to work fine from the phone to *, because I can trigger menus on my own IVR without a problem. I don't specifically recall the minor, but I know the old asterisk distribution major was 1.2. Maybe the minor was high 20s, 30s. Who knows. The new distribution is 1.4.23.1. I upgraded my 7960 from 8.6.0 to 8.11.0 with the transition to the new distro, so perhaps that's an issue. (FTR, my call path is: SIP from the phone (7960) to *, IAX from * to the provider.) Any ideas? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2000 BLF
Never mind, found magic. We have to set account to the line that represents that context in Asterisk. Phone works pretty well for a POE, dual Ethernet, 4 line phone that accepts a 2.5 mm headset, has 6 line display, and all the expected features for $79.95. Speaker phone is clear, $9.95 Panasonic headset works great on it. The worst feature is that it says Made in China on the bottom, and I would rather not send our money to China, but... The one operational thing I don't like is that when a call drops, the phone returns to dial tone rather than hangs up like the SNOM does. But, other features are good. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Tuesday, March 17, 2009 7:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Grandstream GXP2000 BLF We have a system running SNOM 360s, and BLF works fine. We are trying Grandstream GXP2000s and like the phones for what they are, but can't get the BLF to work. The IB just says to set to BLF and put in the phone number. We have tried variations like adding @xxx.xxx.yyy.zzz, but no lights light. Does anyone have the magic incantation to get the BLF to work? Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Vlasis Hatzistavrou (KTI) wrote: Fax transmission can be initiated from any one of the parties. AFAIK T.38 as well as the PSTN fax standards do not show any preference whether fax transmission is requested from a or b party. In practice, the caller usually initiates a fax transmission, but this doesn't mean that the called party cannot initiate it, too. Best regards, Vlasis Hatzistavrou. Steve Underwood wrote: Hey, why bother looking at a spec when its so much more fun to make it up as we go along? ... Regards, Steve I don't think there is a need to be ironic here... I wrote AFAIK which we all know means as far as I know, so why the bashing? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wideband g711-HD vs. g711.1?
Hi! has anyone seen specifications of the codec g711-HD? This is right now spreading fast in the wake up CATiq (the DECT successor), for example in the AVM products (www.avm.de). Googling for G.711-HD only produces hits about AVM. The AVM web site is very vague. AVM support answered: g711-HD is g711 A-Law sampled with 16 kHz. Currently AVM does not have intentions to support Siren7, Siren14, SILK or CELT in the near future, they will stick to g722 and (g711-HD between their own devices with double the bandwidth of g722 when this is readily available). G.711.1 is a really brain dead codec. I find it hard to believe there will ever be much take up of it. Still I am curious: What exactly is braindead about it? Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2000 BLF
On Tue, 17 Mar 2009, Cary Fitch wrote: Never mind, found magic. We have to set account to the line that represents that context in Asterisk. thread hijack, but never mind... Phone works pretty well for a POE, dual Ethernet, 4 line phone that accepts a 2.5 mm headset, has 6 line display, and all the expected features for $79.95. Speaker phone is clear, $9.95 Panasonic headset works great on it. The worst feature is that it says Made in China on the bottom, and I would rather not send our money to China, but... What phones aren't made in China these days? The one operational thing I don't like is that when a call drops, the phone returns to dial tone rather than hangs up like the SNOM does. But, other features are good. On each account page, near the bottom there is an option: Turn off speaker on remote disconnect: Set this to yes and you'll have your wish. Gordon Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Tuesday, March 17, 2009 7:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Grandstream GXP2000 BLF We have a system running SNOM 360s, and BLF works fine. We are trying Grandstream GXP2000s and like the phones for what they are, but can't get the BLF to work. The IB just says to set to BLF and put in the phone number. We have tried variations like adding @xxx.xxx.yyy.zzz, but no lights light. Does anyone have the magic incantation to get the BLF to work? Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
2009/3/17 Steve Underwood ste...@coppice.org Vlasis Hatzistavrou (KTI) wrote: Olivier wrote: Hi, I've been playing with T.38. I observed that mostly but not always, it's the calling endpoint that reINVITE the other party to drop current SIP/G711 session and start a new T.38. But sometimes, it's also the callee party that reINVITE the calling party. Which is the standardized or most common, way to start a T.38 session ? Shall it come from callee or from caller ? Regards Fax transmission can be initiated from any one of the parties. AFAIK T.38 as well as the PSTN fax standards do not show any preference whether fax transmission is requested from a or b party. In practice, the caller usually initiates a fax transmission, but this doesn't mean that the called party cannot initiate it, too. Best regards, Vlasis Hatzistavrou. Hey, why bother looking at a spec when its so much more fun to make it up as we go along? T.38 says that if the call starts in audio mode it is the called end which should initiate a re-invite to change from audio to T.38. This makes sense, as that is the end which has the best chance of figuring out if a FAX machine answers the call. In practice many T.38 implementations will send out a re-invite when they are the calling side, so any practical implementation has to allow for this. Clashes are possible, if both ends send re-invite, and this is not always handled properly Yesterday, with 2 consecutive sendings on the same setup (same fax file, same ATAs, same servers), on the first try, I've seen the reINVITE coming from callee on from the caller on the second try. I don't remember I changed anything between both tries (though I may have done without noticing this). Also many implementations will only listen for a FAX machine at the beginning of a call, so if a human answers and later presses the start button on their FAX machine the T.38 gateway might miss this. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
2009/3/17 Vlasis Hatzistavrou (KTI) vh...@kinetix.gr Vlasis Hatzistavrou (KTI) wrote: Fax transmission can be initiated from any one of the parties. AFAIK T.38 as well as the PSTN fax standards do not show any preference whether fax transmission is requested from a or b party. In practice, the caller usually initiates a fax transmission, but this doesn't mean that the called party cannot initiate it, too. Best regards, Vlasis Hatzistavrou. Steve Underwood wrote: Hey, why bother looking at a spec when its so much more fun to make it up as we go along? ... Regards, Steve I don't think there is a need to be ironic here... I wrote AFAIK which we all know means as far as I know, so why the bashing? Vlasis, I don't think Steve's irony where targeted to you but to those which are supposed to read specs (ATA vendors) ... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Olivier wrote: 2009/3/17 Vlasis Hatzistavrou (KTI) vh...@kinetix.gr mailto:vh...@kinetix.gr Vlasis Hatzistavrou (KTI) wrote: Fax transmission can be initiated from any one of the parties. AFAIK T.38 as well as the PSTN fax standards do not show any preference whether fax transmission is requested from a or b party. In practice, the caller usually initiates a fax transmission, but this doesn't mean that the called party cannot initiate it, too. Best regards, Vlasis Hatzistavrou. Steve Underwood wrote: Hey, why bother looking at a spec when its so much more fun to make it up as we go along? ... Regards, Steve I don't think there is a need to be ironic here... I wrote AFAIK which we all know means as far as I know, so why the bashing? Vlasis, I don't think Steve's irony where targeted to you but to those which are supposed to read specs (ATA vendors) ... Hello Olivier, Well, since Steve's comment followed right after my reply, it seemed like the comment was very much targeted at me... The comment can be taken both ways I guess... Regards, Vlasis. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA3102 - How to save config in a file
Hi, I've read in this mailinglist archives some notes related to Linksys SPA3102 provisioning but I couldn't find there the answer I'm looking for. Is it possible with this box (mine is unlocked) to store its config file(s) in a TFTP server, and have this(these) file(s) reloaded at boot time, for instance ? In embedded web server, there is a Provisioning tab full of settings but none seems to fit. Any hint ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2000 BLF
Well yeah, even the SNOMs are Engineered in Germany, made in China. And thanks for the tip on Speaker drop. It actually is Line drop rather than only Speaker drop, but it works fine. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Tuesday, March 17, 2009 8:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Grandstream GXP2000 BLF On Tue, 17 Mar 2009, Cary Fitch wrote: Never mind, found magic. We have to set account to the line that represents that context in Asterisk. thread hijack, but never mind... Phone works pretty well for a POE, dual Ethernet, 4 line phone that accepts a 2.5 mm headset, has 6 line display, and all the expected features for $79.95. Speaker phone is clear, $9.95 Panasonic headset works great on it. The worst feature is that it says Made in China on the bottom, and I would rather not send our money to China, but... What phones aren't made in China these days? The one operational thing I don't like is that when a call drops, the phone returns to dial tone rather than hangs up like the SNOM does. But, other features are good. On each account page, near the bottom there is an option: Turn off speaker on remote disconnect: Set this to yes and you'll have your wish. Gordon Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Tuesday, March 17, 2009 7:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Grandstream GXP2000 BLF We have a system running SNOM 360s, and BLF works fine. We are trying Grandstream GXP2000s and like the phones for what they are, but can't get the BLF to work. The IB just says to set to BLF and put in the phone number. We have tried variations like adding @xxx.xxx.yyy.zzz, but no lights light. Does anyone have the magic incantation to get the BLF to work? Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 - How to save config in a file
I'm not sure how this work with Linksys, but with Polycom, you just touch a file in the TFTP directory (syncinfo.xml), and this causes the phone to do it's file transfers on reboot. Could be a Polycom thing, but I'd bet there's a fair chance that they work similarly. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Tuesday, March 17, 2009 8:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SPA3102 - How to save config in a file Hi, I've read in this mailinglist archives some notes related to Linksys SPA3102 provisioning but I couldn't find there the answer I'm looking for. Is it possible with this box (mine is unlocked) to store its config file(s) in a TFTP server, and have this(these) file(s) reloaded at boot time, for instance ? In embedded web server, there is a Provisioning tab full of settings but none seems to fit. Any hint ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PBX to gate interface
Has anyone found a good wayt o do a gate intercom using Asterisk? I am looking at a Xorcom PBX with programmable contact, so I have no issue with opening the gate, but the interface at the gate is a bit tricky. I thought about a weather proof housing containing a phone but it seems a bit tacky. I also looked at a handsfree erather proof phone, but at $600 it is a bit steep. Any solutions that have been implemented successfully? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Olivier wrote: 2009/3/17 Vlasis Hatzistavrou (KTI) vh...@kinetix.gr mailto:vh...@kinetix.gr Vlasis Hatzistavrou (KTI) wrote: Fax transmission can be initiated from any one of the parties. AFAIK T.38 as well as the PSTN fax standards do not show any preference whether fax transmission is requested from a or b party. In practice, the caller usually initiates a fax transmission, but this doesn't mean that the called party cannot initiate it, too. Best regards, Vlasis Hatzistavrou. Steve Underwood wrote: Hey, why bother looking at a spec when its so much more fun to make it up as we go along? ... Regards, Steve I don't think there is a need to be ironic here... I wrote AFAIK which we all know means as far as I know, so why the bashing? Vlasis, I don't think Steve's irony where targeted to you but to those which are supposed to read specs (ATA vendors) Oh, it was meant for him. In the time it took him to write his wrong e-mail he could have gone to the ITU web site, downloaded a free copy of the T.38 spec, looked up the annex where it described the negotiation process, and found a clear statement of what is supposed to happen. Of course, that wouldn't tell him the real world issues, like the fact half the T.38 implementations out there don't follow the spec., but it would have been a valuable start. It would also keep the noise level on this list down. What a lot of people don't allow for when writing garbage is it stays on the internet for years, and eventually becomes reference material. :-\ Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 - How to save config in a file
Hi, The format of the file for the provisioning is xml. You create a file with the configuration you want and put it on your provisioning server. Then, you put a rule in the spa3102 to retrieve the file when the unit boot up. Jimmy -Original Message- From: oza-4...@myamail.com Sent: Tue, 17 Mar 2009 14:13:08 +0100 To: asterisk-users@lists.digium.com Subject: [asterisk-users] SPA3102 - How to save config in a file Hi, I've read in this mailinglist archives some notes related to Linksys SPA3102 provisioning but I couldn't find there the answer I'm looking for. Is it possible with this box (mine is unlocked) to store its config file(s) in a TFTP server, and have this(these) file(s) reloaded at boot time, for instance ? In embedded web server, there is a Provisioning tab full of settings but none seems to fit. Any hint ? Regards GET FREE 5GB EMAIL - Check out spam free email with many cool features! Visit http://www.inbox.com/email to find out more! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Direct Dial-Out and CDR destination numbers
what about relogging the information using:
Set(CDR(customfield)=${CDR(originalfield)})
i think?
who knows, i might be wrong with all of this but i guess it will work...
2009/3/17 Matthias Urlichs matth...@urlichs.de
Hi,
as German phone numbers are variable_length, I need to use direct dial-out.
The problem is that only the part which appears in extensions.ael (and thus
in the argument to Dial()) is logged to the call data record.
What I want, obviously, is for the Dial() app to append the additional
digits to the CDR's destination number.
Is that possible?
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Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Hi Olivier, Olivier wrote: T.38 says that if the call starts in audio mode it is the called end which should initiate a re-invite to change from audio to T.38. This makes sense, as that is the end which has the best chance of figuring out if a FAX machine answers the call. In practice many T.38 implementations will send out a re-invite when they are the calling side, so any practical implementation has to allow for this. Clashes are possible, if both ends send re-invite, and this is not always handled properly Yesterday, with 2 consecutive sendings on the same setup (same fax file, same ATAs, same servers), on the first try, I've seen the reINVITE coming from callee on from the caller on the second try. I don't remember I changed anything between both tries (though I may have done without noticing this). That is what typically what happens when the calling end doesn't obey the spec. It comes down to a race for who initiates the re-invite first. If you are lucky the two ends sort themselves out. If you are unlucky you end up with both ends re-inviting, and you may get a call failure. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 - How to save config in a file
2009/3/17 Danny Nicholas da...@debsinc.com I’m not sure how this work with Linksys, but with Polycom, you just “touch” a file in the TFTP directory (syncinfo.xml), and this causes the phone to do it’s file transfers on reboot. Could be a Polycom thing, but I’d bet there’s a fair chance that they work similarly. Unfortunately, 3102's embedded server is silent about that. I won't be surprised you could upload a config file using DHCP options, but it's not easy to guess all parameters ... I'll open another thread on that, but you meet standalone devices that ideally, should retrieve from a server, a config file when booting and backup config data from time to time ... I don't know if, playing with TFTP, DHCP and http load testers, it's possible to provide auto-provision services to those devices ... -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Tuesday, March 17, 2009 8:13 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] SPA3102 - How to save config in a file Hi, I've read in this mailinglist archives some notes related to Linksys SPA3102 provisioning but I couldn't find there the answer I'm looking for. Is it possible with this box (mine is unlocked) to store its config file(s) in a TFTP server, and have this(these) file(s) reloaded at boot time, for instance ? In embedded web server, there is a Provisioning tab full of settings but none seems to fit. Any hint ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wideband g711-HD vs. g711.1?
Philipp von Klitzing wrote: Hi! has anyone seen specifications of the codec g711-HD? This is right now spreading fast in the wake up CATiq (the DECT successor), for example in the AVM products (www.avm.de). Googling for G.711-HD only produces hits about AVM. The AVM web site is very vague. AVM support answered: g711-HD is g711 A-Law sampled with 16 kHz. Currently AVM does not have intentions to support Siren7, Siren14, SILK or CELT in the near future, they will stick to g722 and (g711-HD between their own devices with double the bandwidth of g722 when this is readily available). G.711.1 is a really brain dead codec. I find it hard to believe there will ever be much take up of it. Still I am curious: What exactly is braindead about it? See for yourself. The spec free, though the ITU patents database lists patents on its methods - http://www.itu.int/rec/T-REC-G.711.1-200803-P/en Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
Marco Sambo schrieb: Anyone know how to use busy-level parameter or some other useful parameters? call-limit=2 busy-level=1 ? busy-level is not in Asterisk 1.4 of course. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] Re: Plastic Water Bottles
Tzafrir Cohen schrieb: Sorry for following on this off-topic, but, On Mon, Mar 16, 2009 at 08:49:53PM -0600, drew einhorn wrote: The plastics industry says polycarbonate bottles are safe. http://www.bisphenol-a.org/about/faq.html#g I'm sure Maggie and here friends would say ALL plastic bottles are very dangerous. This lady seems to be at a reasonable middle ground. http://trusted.md/blog/vreni_gurd/2007/03/29/plastic_water_bottles http://www.cancer.ca/Canada-wide/About cancer/Cancer myths/Reusing disposable water bottles.aspx?sc_lang=en is also relevant. For some strange reason your URL does not work for me. Content not available at this time. http://www.cancer.ca/Canada-wide/About%20cancer/Cancer%20myths/Reusing%20disposable%20water%20bottles.aspx Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 - How to save config in a file
Jimmy Godbout wrote: Hi, The format of the file for the provisioning is xml. You create a file with the configuration you want and put it on your provisioning server. Then, you put a rule in the spa3102 to retrieve the file when the unit boot up. Well, with the other Linksys devices (SPA-941 etc), you can retrieve the configuration from the phone first - I think that's what the OP had in mind. /Per Jessen, Zürich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 - How to save config in a file
Olivier schrieb: Hi, I've read in this mailinglist archives some notes related to Linksys SPA3102 provisioning but I couldn't find there the answer I'm looking for. Is it possible with this box (mine is unlocked) to store its config file(s) in a TFTP server, and have this(these) file(s) reloaded at boot time, for instance ? In embedded web server, there is a Provisioning tab full of settings but none seems to fit. Any hint ? Regards hello, you could retrieve the config from you SPA with the following url: http://ipofyourphone/admin/spacfg.xml . this file could be directly provisionend via tftp, http or ftp by entering the url in the provisioning section or loading a url http://ipofyourphone/admin/resync?http://1.2.3.4/config.xml maybe you need a firmwareupdate before the spacfg.xml could be retrieved, IMO it works only with an newer than 5.x firmware. best regards steve smith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kewlstart - Busy signal before battery drop.
Hello all. I have Asterisk connected to an Adit 600 channel bank with a TE110P and the channel bank is connected to a PBX providing dialtone to the PBX with fxo_ks signalling. When a call between the PBX and Asterisk completes there is a momentary battery drop/reversal or something that signals the PBX that the other side has hung up and then the PBX hangs up. This all works fine. However, when asterisk hangs up it also immediately starts playing a busy signal. My issue is that the busy signal begins playing before the battery drop occurs. This means that at the end of any calls or voicemails on the PBX there is a .5-1 second interval of a busy tone at the end. Is there any way to get the busy tone to begin *after* the battery drop? I've tried messing with the indications.conf file but didn't have any luck and I can't see anything in chan_dahdi.conf or system.conf. This same thing happens at home with my TDM400P so I'm inclined to think it's not exclusive to the channel bank. Anyone have any ideas? Thanks -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra 9133i programmable buttons (* 4.1.23)
The 1.x firmware for Aastra's (for the 9112i / 9133i / 480i) do support some of the XML functionality that you see in the newer 2.x firmware (for the more recent models). I;m not sure if controlling LED status of the keys is supported by 1.x - but you should be able to find that out by taking a look at Aastra's XML API document here: http://www.aastra.com/cps/rde/xbcr/SID-3D8CCB6A-2E5763F4/04/Telecom_PA-001004-00-03_XML_Development_Guide_Release_1.4.2.zip -- Matt On Mon, Mar 16, 2009 at 1:34 PM, Steve Davies davies...@gmail.com wrote: 2009/3/16 David Ruggles da...@safedatausa.com: Is it possible to control the light on a programmable button without the blf option? I'm using a programmable button to turn call recording on and off and I would like the light to indicate the status. Thanks, 9133i phones are pretty much obsolete, and are not getting firmware updates, so I do not know whether Aastra ever put any of their XML application control code into that model. If they did, then it should be possible to respond with button status using XML updates from the server, otherwise you'd need to upgrade to one of their currently supported phones, which are almost certainly capable of this sort of thing. PS. I have never personally used the XML facility of Aastra phones, but I hear quite good things about it. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
When I was first looking at Aastra, over a year ago, I thought there was some talk that Aastra was more supportive of asterisk then most vendors. Is this still true? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: Cory Andrews [mailto:c...@voipsupply.com] Sent: Monday, March 16, 2009 6:37 PM To: da...@safedatausa.com Subject: RE: [asterisk-users] Good phone near $125 David - not sure if you have any specific requirements in terms of # of lines or other features, but the Polycom IP330 and Linksys SPA942 are excellent phones which are in your price range. http://www.voipsupply.com/polycom-ip-330 http://www.voipsupply.com/linksys-spa942 Also the Grandstream GXP2010 and Aastra 6731i http://www.voipsupply.com/grandstream-gxp2010 http://www.voipsupply.com/catalog/product/view/id/7991/s/aastra-6731i-ip-pho ne/ Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Monday, March 16, 2009 6:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Good phone near $125 I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.13/1999 - Release Date: 03/16/09 07:04:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ast/Hyla/IAX Scalability?
While we have your attention Steve (Underwood) do you have a high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We currently use 0.0.4 with a very high success rate. Is there any benefit in moving up to a newer library? I looked at the Changelog in the source, but it stopped at 0.0.4. Thanks for any feedback. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Noisy Ring Back Tone with TE205P card
Hi,
I stilll continue with the problem but I have noticed something new that
maybe a clue. The noise during the call progress is made by the
appearance of the different lines in the asterisk CLI, I mean, each line
is posted in the CLI generates a noise in the call's signallling tone.
For example, if I try doing a call during a resetinterval option, which
reset all free channels, each line posted in the CLI generates a burst
noise.
Any ideas?
Thanks
Regards
Imanol Pardavila escribió:
Hi,
I am having problems with an Asterisk with a Digium TE205P card. The
issue is that the Ring Back Tone is noisy. I am making modem's calls
and this noise influences on the initial negotiation protocol, so
modems have to recall.
My configuration is:
Asterisk version: Asterisk 1.4.21.2
Linux version: CentOS release 5.2 (Final)
Card: Digium TE205P
##zapata.conf#
; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is
intended
; to be #include-d by /etc/zapata.conf that will include the global
settings
;
[channels]
; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1
language=es
context=default
switchtype=euroisdn
pridialplan=unknown
prilocaldialpla=national
signalling=pri_cpe
resetinterval=never
group=1
channel = 1-15,17-31
; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
group=2
channel = 32-46,48-62
##zaptel.conf#
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER)
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
hardhdlc=16
# Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
hardhdlc=47
# Global data
loadzone= es
defaultzone = es
##extensions.conf###
exten =999888777,1,Goto(JUMP,s,1)
[JUMP]
exten = s,1,Dial(Zap/R2/6,15,r)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Goto(HANGUP,s,1)
exten = s-NOANSWER,1,Goto(HANGUP,s,1)
exten = s-CHANUNAVAIL,1,Goto(HANGUP,s,1)
exten = s-CONGESTION,1,Goto(HANGUP,s,1)
[HANGUP]
exten = s,1,Hangup
[DID_span_1]
include = default
[DID_span_2]
include = default
I have no idea about where could be the problem. I can't see anything
rare in the logs
Any ideas?
Thanks
Regards
___
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Re: [asterisk-users] Ast/Hyla/IAX Scalability?
On Tue, Mar 17, 2009 at 10:34 AM, Steve Davies davies...@gmail.com wrote: While we have your attention Steve (Underwood) do you have a high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We currently use 0.0.4 with a very high success rate. Is there any benefit in moving up to a newer library? I looked at the Changelog in the source, but it stopped at 0.0.4. I'm not Steve, but I can tell you what I've found. 0.0.5 seems to be required for building asterisk-1.6.0.6 0.0.6 introduced some API changes, and trunk has been updated, but asterisk-1.6.0.6 does not have those changes, and you won't be able to compile unless you replace app_fax.c with the version in the 1.6.0 subversion branch. I suspect (since it's in 1.6.0 branch) but cannot confirm that the next 1.6.0. release will have the changes to build / use spandsp-0.0.6 the 5 and 6 have progressively better support for misbehaving faxes, whereas faxes that behave according to spec work well on all versions I've tried. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 - How to save config in a file
2009/3/17 Stefan Schmidt s...@sil.at Olivier schrieb: Hi, I've read in this mailinglist archives some notes related to Linksys SPA3102 provisioning but I couldn't find there the answer I'm looking for. Is it possible with this box (mine is unlocked) to store its config file(s) in a TFTP server, and have this(these) file(s) reloaded at boot time, for instance ? In embedded web server, there is a Provisioning tab full of settings but none seems to fit. Any hint ? Regards hello, you could retrieve the config from you SPA with the following url: http://ipofyourphone/admin/spacfg.xml . this file could be directly provisionend via tftp, http or ftp by entering the url in the provisioning section or loading a url http://ipofyourphone/admin/resync?http://1.2.3.4/config.xml maybe you need a firmwareupdate before the spacfg.xml could be retrieved, IMO it works only with an newer than 5.x firmware. best regards steve smith Thanks : that's exactly what I was looking for ! I'll try it later today and report back here. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
Ok, I read it. Thank u. For busy on SIP I use also the Asterisk peer function SIPPEER with field CURCALLS. 2009/3/17 Philipp Kempgen philipp.kemp...@amooma.de Marco Sambo schrieb: Anyone know how to use busy-level parameter or some other useful parameters? call-limit=2 busy-level=1 ? busy-level is not in Asterisk 1.4 of course. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ast/Hyla/IAX Scalability?
David Backeberg wrote: On Tue, Mar 17, 2009 at 10:34 AM, Steve Davies davies...@gmail.com wrote: While we have your attention Steve (Underwood) do you have a high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We currently use 0.0.4 with a very high success rate. Is there any benefit in moving up to a newer library? I looked at the Changelog in the source, but it stopped at 0.0.4. I'm not Steve, but I can tell you what I've found. 0.0.5 seems to be required for building asterisk-1.6.0.6 0.0.6 introduced some API changes, and trunk has been updated, but asterisk-1.6.0.6 does not have those changes, and you won't be able to compile unless you replace app_fax.c with the version in the 1.6.0 subversion branch. I suspect (since it's in 1.6.0 branch) but cannot confirm that the next 1.6.0. release will have the changes to build / use spandsp-0.0.6 the 5 and 6 have progressively better support for misbehaving faxes, whereas faxes that behave according to spec work well on all versions I've tried. That's about right. The changes to make 0.0.6pre6 and beyond work with Asterisk 1.6.0 are tiny - well, you can see them in 1.6.1, as it uses the date stamp of the spandsp version to select how to build. quite old versions of spandsp work well with clean FAXes. spandsp-0.0.6pre7 works well with some pretty messed up ones. I believe the results with 0.0.6pre7 should be comparable to spandsp + iaxmodem + HylaFAX, as long as the timing within your Asterisk is OK. The use of IAX introduces the possibility of dropped packets, but also adds some timing elasticity. For the kind of results you can expect see http://www.soft-switch.org/spandsp-soft-fax-performance.html If you have more than 1% FAX failures which cannot be explained, you have a problem :-) Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ast/Hyla/IAX Scalability?
2009/3/17 David Backeberg dbackeb...@gmail.com: On Tue, Mar 17, 2009 at 10:34 AM, Steve Davies davies...@gmail.com wrote: While we have your attention Steve (Underwood) do you have a high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We currently use 0.0.4 with a very high success rate. Is there any benefit in moving up to a newer library? I looked at the Changelog in the source, but it stopped at 0.0.4. I'm not Steve, but I can tell you what I've found. 0.0.5 seems to be required for building asterisk-1.6.0.6 0.0.6 introduced some API changes, and trunk has been updated, but asterisk-1.6.0.6 does not have those changes, and you won't be able to compile unless you replace app_fax.c with the version in the 1.6.0 subversion branch. I suspect (since it's in 1.6.0 branch) but cannot confirm that the next 1.6.0. release will have the changes to build / use spandsp-0.0.6 the 5 and 6 have progressively better support for misbehaving faxes, whereas faxes that behave according to spec work well on all versions I've tried. Thank you - a useful summary. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
At 01:29 AM 3/17/2009, you wrote: But there is another little problem. On Aastra phone (on other phones I don't try yet), the xfer button doesn't work, until I set call-limit=2, but making this I find the phone not busy. As far as I can tell on my Aastra phones it takes 2 links to complete a transfer. Pressing transfer puts the first call on hold and allows you to make a second call. Pressing transfer a second time then connects those to calls together and removes you from the call. If you only have 1 call allowed you'll need to implement that using features.conf or implement the busy stuff in the dial plan. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with largeuser base?
On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: Hi Yehavi, Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time. Regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
Polycom IP 430 or 330. asterisk-users-requ...@lists.digium.com wrote: Date: Mon, 16 Mar 2009 18:24:33 -0400 From: David Ruggles da...@safedatausa.com I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with largeuser base?
On Tue, 17 Mar 2009 10:00:56 -0700 (PDT), Vincent Li wrote: On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: Hi Yehavi, Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time. There was at Sam Houston Stat University in Texas, but they have since transitioned to a Cisco Call Manager system...essentially reversing their earlier migration. I gather that this decision was driven by changes in their staffing and epecially the loss of key staff knowledgable in the ways of Asterisk. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with largeuser base?
2009/3/17 Michael Graves mgra...@mstvp.com On Tue, 17 Mar 2009 10:00:56 -0700 (PDT), Vincent Li wrote: On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: Hi Yehavi, Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time. There was at Sam Houston Stat University in Texas, but they have since transitioned to a Cisco Call Manager system...essentially reversing their earlier migration. I gather that this decision was driven by changes in their staffing and epecially the loss of key staff knowledgable in the ways of Asterisk. Are those staffing changes the consequence of issues in Asterisk deployment or is it the opposite (the new staff members that decided to change back to CCM) ? Given the cost of reverting to CCM, that would be strange Sam Houston Stat University in Texas prefers to roll back to CCM instead of finding appropriate support suppliers. Maybe actors are still reading this list and could tell more about it. I know these days, it's easier to get hudge bargains from vendors ... Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com sip%3amgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with large user base?
On Mon, Mar 16, 2009 at 5:34 PM, Vincent Li vincent.mc...@gmail.com wrote: Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. http://www.networkworld.com/news/2007/071707-open-source-voip.html http://www.digium.com/en/company/casestudies/viewcasestudies/University-of-Pennsylvania Those links gets passed around every time this topic comes up. I don't know what metrics led to the conclusion of the project manager, nor the way things were configured in your particular pilot. Asterisk-1.6 has dramatically enhanced SIP handling compared to 1.4. It also has dramatically faster large-dialplan handling. You can read all about it in the files that come packaged with 1.6. It's possible (I would dare say likely) that the project manager is looking at old data, or that the pilot was done with old versions of asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with largeuser base?
Hi Visit, that's not correct - google Sam Houston University It's a pretty well known asterisk installation. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vincent Li Sent: Tuesday, March 17, 2009 1:01 PM To: Yehavi Bourvine Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk is not designed for University with largeuser base? On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: Hi Yehavi, Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time. Regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with largeuser base?
Very complex installation, so try to star with: 1) Compatibility of current phone platform + asterisk. For example, you can convert current extension as sip extension using fxs ports. This reduces your cost, you don't need buy 8.000 ip phones and install an new wired network. 2) Planning and do an asterisk cluster based building an locations. Group extensions by buildings/asterisk servers. 4) Planning and do asterisk network with and distributed dial plan and trunking 5) Try locate an asterisk specialists 6) believe in asterisk! Regards, Luis Morales On Tue, Mar 17, 2009 at 12:46 PM, Dean Collins d...@cognation.net wrote: Hi Visit, that's not correct - google Sam Houston University It's a pretty well known asterisk installation. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vincent Li Sent: Tuesday, March 17, 2009 1:01 PM To: Yehavi Bourvine Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk is not designed for University with largeuser base? On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: Hi Yehavi, Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time. Regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI or Zaptel doesn't compile against 1.4.24
Hi, We installed the latest 1.4.24 on a test machine and can't get zaptel nor dahdi compile. It's a Linux Debian Etch. Errors we have: keewi:/usr/src/dahdi-linux-2.1.0.4# make make -C /lib/modules/2.6.18-custom.2/build ARCH=i386 SUBDIRS=/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.1.0.4/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. CC [M] /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.o In file included from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:38: /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:23:27: error: linux/version.h: Aucun fichier ou répertoire de ce type /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:5: warning: LINUX_VERSION_CODE is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:26: warning: KERNEL_VERSION is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:40: error: missing binary operator before token ( /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:5: warning: LINUX_VERSION_CODE is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:27: warning: KERNEL_VERSION is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:41: error: missing binary operator before token ( In file included from include/linux/kernel.h:11, from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:40: include/linux/linkage.h:4:25: error: asm/linkage.h: Aucun fichier ou répertoire de ce type In file included from include/linux/posix_types.h:47, from include/linux/types.h:14, from include/linux/kernel.h:13, from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:40: /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:13:22: error: features.h: Aucun fichier ou répertoire de ce type aso. Zaptel, the same: ... make[1]: entrant dans le répertoire « /usr/src/asterisk-1.4.24/zaptel » make -C /lib/modules/2.6.18-custom.2/build SUBDIRS=/usr/src/asterisk-1.4.24/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M=wcfxo.o zaptel.o ztdummy.o zttranscode.o modules make[2]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. CC [M] /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.o In file included from include/linux/kernel.h:11, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: include/linux/linkage.h:4:25: error: asm/linkage.h: Aucun fichier ou répertoire de ce type In file included from include/linux/posix_types.h:47, from include/linux/types.h:14, from include/linux/kernel.h:13, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:13:22: error: features.h: Aucun fichier ou répertoire de ce type /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:14:35: error: no include path in which to search for asm/posix_types.h In file included from include/linux/kernel.h:13, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: include/linux/types.h:15:23: error: asm/types.h: Aucun fichier ou répertoire de ce type aso. What are we doing wrong? -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mobile centrex solution
Two of the wireless carriers have a Centrex-like solution: http://www.networkcomputing.com/channels/wireless/showArticle.jhtml?articleI D=202200832pgno=5 Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Fort Sent: Tuesday, March 17, 2009 3:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] mobile centrex solution anyone know of a solution where mobile handsets out roaming the pstn cellular network can be used and treated as full fleged centrex extentions, i.e. I can transfer a call that comes in on a wired centrex copper pair out to a cell phone and the cell phone can transfer the call back or vice versa where the cell phone recieves the call directly and can transfer to the office all without hairpinning the call? essentially when the call is transfered I'd like to have asterisk get out of the call path but still have the capability to transfer the call back to asterisk and it's attached office phones. Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI or Zaptel doesn't compile against 1.4.24
make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. specifically Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing Are you using the stock Debian kernel? If so, do you have the linux kernel source and kernel headers source package installed? If so, make sure the source packages installed are the same version number of the current running kernel. -John Knight Administrator TOOTAI wrote: Hi, We installed the latest 1.4.24 on a test machine and can't get zaptel nor dahdi compile. It's a Linux Debian Etch. Errors we have: keewi:/usr/src/dahdi-linux-2.1.0.4# make make -C /lib/modules/2.6.18-custom.2/build ARCH=i386 SUBDIRS=/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.1.0.4/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. CC [M] /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.o In file included from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:38: /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:23:27: error: linux/version.h: Aucun fichier ou répertoire de ce type /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:5: warning: LINUX_VERSION_CODE is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:26: warning: KERNEL_VERSION is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:40: error: missing binary operator before token ( /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:5: warning: LINUX_VERSION_CODE is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:27: warning: KERNEL_VERSION is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:41: error: missing binary operator before token ( In file included from include/linux/kernel.h:11, from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:40: include/linux/linkage.h:4:25: error: asm/linkage.h: Aucun fichier ou répertoire de ce type In file included from include/linux/posix_types.h:47, from include/linux/types.h:14, from include/linux/kernel.h:13, from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:40: /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:13:22: error: features.h: Aucun fichier ou répertoire de ce type aso. Zaptel, the same: ... make[1]: entrant dans le répertoire « /usr/src/asterisk-1.4.24/zaptel » make -C /lib/modules/2.6.18-custom.2/build SUBDIRS=/usr/src/asterisk-1.4.24/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M=wcfxo.o zaptel.o ztdummy.o zttranscode.o modules make[2]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. CC [M] /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.o In file included from include/linux/kernel.h:11, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: include/linux/linkage.h:4:25: error: asm/linkage.h: Aucun fichier ou répertoire de ce type In file included from include/linux/posix_types.h:47, from include/linux/types.h:14, from include/linux/kernel.h:13, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:13:22: error: features.h: Aucun fichier ou répertoire de ce type /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:14:35: error: no include path in which to search for asm/posix_types.h In file included from include/linux/kernel.h:13, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: include/linux/types.h:15:23: error: asm/types.h: Aucun fichier ou répertoire de ce type aso. What are we doing wrong? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with largeuser base?
See too: http://www.networkworld.com/news/2007/011907-mit-your-take.html?page=1 Jorge Mendoza Dean Collins wrote: Hi Visit, that's not correct - google Sam Houston University It's a pretty well known asterisk installation. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vincent Li Sent: Tuesday, March 17, 2009 1:01 PM To: Yehavi Bourvine Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk is not designed for University with largeuser base? On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: Hi Yehavi, Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time. Regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
A2billing is a good fit for that then. Yeah, voipon. Thanks for the input Gordon. Maybe worth hooking up offline if we're doing similar stuff. Gavin. On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote: On Tue, 17 Mar 2009, Gavin Henry wrote: 2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Mon, 16 Mar 2009, Gavin Henry wrote: When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. Yeah, don't really like them though. I could go down to a 51i for £67 ex VAT. Grandstreams aren't to everyones liking, this is true... You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. What would that come in at? A Dell T100 is £300 ex VAT for 160GB, 1GB RAM and a Dual Core Intel® Pentium® E2220; 2.4GHz with 3yrs nxt bday. Under £200 from someone like http://linitx.com/ I don't put disk drives in my boxes though - they boot out of flash. I guess with the Dell, you have on-site or next day replacement if you take that deal though. A 4 port FXO card is £126.95 ex vat. (From voipon by the looks of that price ;-) Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! Thanks. I've been approcached by a client who wants a sort of hotel billing system though - tailored to their needs - it's for a retirement home sort of thing. I suggested they just did a fixed-price deal with the inmates, but that didn't go down well. They want to account for everything to the last penny )-: I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Yeah, as I planned, but not for this project. Good luck! Gordon -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
Yeah, I've experienced that. But what can you do other than stick woth a fat codec. On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote: On Tue, 17 Mar 2009, Gavin Henry wrote: 2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Tue, 17 Mar 2009, Geraint Lee wrote: I know of a local company who're regularly putting 20 concurrent calls over the same broadband setup using G729... Yeah, we use g.729 ourselves too. The issues I've had have been when theres transcoding going on that you can't control - ie. outside your network, so I can go point to point from end-user phone to the people I peer with, but if they then transcode to G711 to go to the PSTN, it's OK, but if it then gets transcoded to GSM for a mobile, or back to G729 to go to an expensive overseas location, then quality does suffer )-: Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI or Zaptel doesn't compile against 1.4.24
If for whatever reason your kernel headers have been corrupted or there is a new version for your particular kernel version, I would suggest purging the package and pulling in the package from the repo -John Knight John Knight wrote: make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. specifically Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing Are you using the stock Debian kernel? If so, do you have the linux kernel source and kernel headers source package installed? If so, make sure the source packages installed are the same version number of the current running kernel. -John Knight Administrator TOOTAI wrote: Hi, We installed the latest 1.4.24 on a test machine and can't get zaptel nor dahdi compile. It's a Linux Debian Etch. Errors we have: keewi:/usr/src/dahdi-linux-2.1.0.4# make make -C /lib/modules/2.6.18-custom.2/build ARCH=i386 SUBDIRS=/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.1.0.4/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. CC [M] /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.o In file included from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:38: /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:23:27: error: linux/version.h: Aucun fichier ou répertoire de ce type /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:5: warning: LINUX_VERSION_CODE is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:26: warning: KERNEL_VERSION is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:40: error: missing binary operator before token ( /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:5: warning: LINUX_VERSION_CODE is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:27: warning: KERNEL_VERSION is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:41: error: missing binary operator before token ( In file included from include/linux/kernel.h:11, from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:40: include/linux/linkage.h:4:25: error: asm/linkage.h: Aucun fichier ou répertoire de ce type In file included from include/linux/posix_types.h:47, from include/linux/types.h:14, from include/linux/kernel.h:13, from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:40: /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:13:22: error: features.h: Aucun fichier ou répertoire de ce type aso. Zaptel, the same: ... make[1]: entrant dans le répertoire « /usr/src/asterisk-1.4.24/zaptel » make -C /lib/modules/2.6.18-custom.2/build SUBDIRS=/usr/src/asterisk-1.4.24/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M=wcfxo.o zaptel.o ztdummy.o zttranscode.o modules make[2]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. CC [M] /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.o In file included from include/linux/kernel.h:11, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: include/linux/linkage.h:4:25: error: asm/linkage.h: Aucun fichier ou répertoire de ce type In file included from include/linux/posix_types.h:47, from include/linux/types.h:14, from include/linux/kernel.h:13, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:13:22: error: features.h: Aucun fichier ou répertoire de ce type /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:14:35: error: no include path in which to search for asm/posix_types.h In file included from include/linux/kernel.h:13, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: include/linux/types.h:15:23: error: asm/types.h: Aucun fichier ou répertoire de ce type aso. What are we doing wrong? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or
Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Steve Underwood wrote: Oh, it was meant for him. In the time it took him to write his wrong e-mail he could have gone to the ITU web site, downloaded a free copy of the T.38 spec, looked up the annex where it described the negotiation process, and found a clear statement of what is supposed to happen. Of course, that wouldn't tell him the real world issues, like the fact half the T.38 implementations out there don't follow the spec., but it would have been a valuable start. It would also keep the noise level on this list down. What a lot of people don't allow for when writing garbage is it stays on the internet for years, and eventually becomes reference material. :-\ Regards, Steve Does AFAIK mean anything at all to you? I never implied that I am the ultimate authority on fax. It has been many years since I read T38 or any other fax specs and apparently I don't remember them to the letter (hence the AFAIK in my sentence). Reference material? Really? My reply on a mailing list can hardly be mistaken for an ITU spec. The fact that my email will remain on the internet for years cannot justify your obnoxious behavior either, unless you honestly believe that my post will misguide the future generations of VoIP implementors for years to come... In other words, if you really wanted to correct my mistake you could have just said that I was wrong. I would even have thanked you for pointing out my error. In such a scenario you would have really contributed against the noise on this list. But unfortunately, all you did was come out as just another wise-guy who desperately needs to get off his high horse. Cheers, Vlasis. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI or Zaptel doesn't compile against 1.4.24
On Tue, Mar 17, 2009 at 07:16:26PM +0100, Administrator TOOTAI wrote: Hi, We installed the latest 1.4.24 on a test machine and can't get zaptel nor dahdi compile. It's a Linux Debian Etch. Errors we have: keewi:/usr/src/dahdi-linux-2.1.0.4# make make -C /lib/modules/2.6.18-custom.2/build ARCH=i386 SUBDIRS=/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.1.0.4/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. CC [M] /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.o In file included from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:38: /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:23:27: error: linux/version.h: Aucun fichier ou répertoire de ce type This is plain wrong. Your source tree is bad. What kernel version do you want to build dahdi against? What kernel version do you use? uname -a /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:5: warning: LINUX_VERSION_CODE is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:26: warning: KERNEL_VERSION is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:24:40: error: missing binary operator before token ( /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:5: warning: LINUX_VERSION_CODE is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:27: warning: KERNEL_VERSION is not defined /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:82:41: error: missing binary operator before token ( In file included from include/linux/kernel.h:11, from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:40: include/linux/linkage.h:4:25: error: asm/linkage.h: Aucun fichier ou répertoire de ce type In file included from include/linux/posix_types.h:47, from include/linux/types.h:14, from include/linux/kernel.h:13, from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:40: /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:13:22: error: features.h: Aucun fichier ou répertoire de ce type aso. Zaptel, the same: ... make[1]: entrant dans le répertoire « /usr/src/asterisk-1.4.24/zaptel » make -C /lib/modules/2.6.18-custom.2/build SUBDIRS=/usr/src/asterisk-1.4.24/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M=wcfxo.o zaptel.o ztdummy.o zttranscode.o modules make[2]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. CC [M] /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.o In file included from include/linux/kernel.h:11, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: include/linux/linkage.h:4:25: error: asm/linkage.h: Aucun fichier ou répertoire de ce type In file included from include/linux/posix_types.h:47, from include/linux/types.h:14, from include/linux/kernel.h:13, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:13:22: error: features.h: Aucun fichier ou répertoire de ce type /usr/lib/gcc/i486-linux-gnu/4.1.2/include/asm/posix_types.h:14:35: error: no include path in which to search for asm/posix_types.h In file included from include/linux/kernel.h:13, from /usr/src/asterisk-1.4.24/zaptel/kernel/wcfxo.c:27: include/linux/types.h:15:23: error: asm/types.h: Aucun fichier ou répertoire de ce type aso. What are we doing wrong? -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 - How to save config in a file
I thought I should also share this : http://www.opensky.ca/~jdhildeb/software/spaconf/ Has anyone tried ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
For MT check out Thirdlane's MT PBX: http://www.thirdlane.com/products/thirdlane-pbx-mte I use the PBX Manager which it's based on and it works very well. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
I finally got around to updating my dialplan to use the new way of doing
callback queues. It seems to me that if one used something like
${CUT(CHANNEL,-,1)} instead of SIP/${EXTEN:3} in the AddQueueMemeber then
the device state of the device the agent logged in from, likely where you
want to call them back at, will be used.
Wouldn¹t this do a better job then assuming the agent logged in from a SIP
user that is the same number as the agent number?
This is what I am using.
; This is used to log on and off agents
exten = *20,1,Answer()
exten = *20,n,wait(.0.5)
exten = *20,n,Read(AgentNumber,agent-user)
exten = *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten = *20,n,GotoIf($[${UserID}=]?NOUSER)
exten = *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)})
exten = *20,n,GotoIf($[${AgentStatus}=1]?VERIFY)
exten = *20,n,GotoIf($[${AgentStatus}=2]?VERIFY)
exten = *20,n(NOUSER),Playback(cfmc/bad-agent)
exten = *20,n,Hangup()
exten = *20,n(VERIFY),VMAuthenticate(${agentnumb...@ourvm)
exten = *20,n,GotoIf($[${AgentStatus}=2]?AGENTOFF)
exten = *20,n,Set(DB(users/${UserID}/AgentStatus)=2)
exten = *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)})
exten =
*20,n,AddQueueMember(support,Local/queue${agentnumb...@ansqueue${CUT(CHA
NNEL,-,1)})
exten = *20,n,Playback(agent-loginok)
exten = *20,n,HangUp()
exten = *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1)
exten = *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)})
exten = *20,n,RemoveQueueMember(support,Local/queue${agentnumb...@ansqueue)
exten = *20,n,Playback(agent-loggedoff)
exten = *20,n,HangUp()
; This is used to call an agent from the queue
exten = _Queue.,1,Set(AgentNumber=${EXTEN:5})
exten = _Queue.,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten =
_Queue.,n,GotoIf($[${DEVICE_STATE(${DB(users/${UserID}/AgentDevice)})}=BU
SY]?ISBUSY)
exten =
_Queue.,n,GotoIf($[${GROUP_COUNT(${user...@phoneinfo)}=0]?DIALIT)
exten = _Queue.,n(ISBUSY),Busy()
exten = _Queue.,n(DIALIT),Set(outbound_group=${user...@phoneinfo)
exten = _Queue.,n,Dial(${DB(users/${UserID}/AgentDevice)},,g)
exten = _Queue.,n,HangUp()
; This is the extension call to get a support agent
exten = 201,1,Answer()
exten = 201,n,Wait(0.5)
exten = 201,n,Set(qac=${QUEUE_MEMBER(support,free)})
exten = 201,n,GotoIf($[${qac} 0]?HAVEAGNT)
exten = 201,n,Playback(cfmc/support-no-agent)
exten = 201,n,Voicemail(2...@ourvm,u)
exten = 201,n,Playback(goodbye)
exten = 201,n,Hangup()
exten = 201,n(HAVEAGNT),Playback(cfmc/support-intro)
exten = 201,n,Queue(support,nrt,,,120)
exten = 201,n,Voicemail(2...@ourvm,b)
exten = 201,n,Playback(goodbye)
exten = 201,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Mark Michelson mmichel...@digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Mon, 09 Mar 2009 14:39:58 -0500
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fwd: add a new queue strategy: SBR
nik600 wrote:
On Mon, Mar 9, 2009 at 3:16 PM, James Sneeringer jsnee...@gmail.com wrote:
If you are using dynamic queues with Local channels (as described in
doc/queues-with-callback-members.txt in the Asterisk source), you can
also optionally implement this functionality directly in the dialplan.
This has the added benefit of allowing you to choose on a per-agent
basis who is eligible for autopause.
-James
thanks for your reply, infact i've implemented the agents in the
dialplan as explained in queues-with-callback-members.txt but this
approach doesn't manage the status of the agent!
I can add / remove / pause / unpause the member interface but what
about the in use status?
The extension in the context will be every time Not in use or shall
i implement hints?
Here there is a piece of my extensions.conf:
[default]
; login procedure for queue 001
exten = _001,1,Answer
exten = _001,n,AddQueueMember(001,Local/${EXTEN:3...@agents)
exten = _001,n,Set(DB(agents/${EXTEN:3})=SIP/${CALLERID(num)})
[agents]
exten = _,hint,${DB(agents/${EXTEN})}
exten = _,1,Dial(${DB(agents/${EXTEN})})
and there isn't an agent but only an extension on a queue.
What do you think about that?
maybe i should open a new post but i think that this kind of approach
isn't much better than the callback functionality, what do you think
about that?
The reason that the member always appears to be not in use is that local
channels are optimized away once they are bridged to their real destination.
The
result of this is that since the channel does not exist anymore, the device
state engine interprets the interface to be not in use anymore. One way to
handle this issue is to change your AddQueueMember call to use
Local/${EXTEN:3...@agents/n (notice the /n at the end). The /n tells the local
channel driver to not attempt to optimize the local channel away.
If
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
On Tue, 17 Mar 2009, Gavin Henry wrote: Yeah, I've experienced that. But what can you do other than stick woth a fat codec. It's tricky. I've been experimenting looking at the possibilitys of using different codecs based on destination, so UK landlines stick to g729 as teh transcode to alaw is OK, but to offshore destiantions look at taking the call in G711... Tricky to get it right without transcoding yourself which you always wnt to avoice (well I do!) Gordon On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote: On Tue, 17 Mar 2009, Gavin Henry wrote: 2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Tue, 17 Mar 2009, Geraint Lee wrote: I know of a local company who're regularly putting 20 concurrent calls over the same broadband setup using G729... Yeah, we use g.729 ourselves too. The issues I've had have been when theres transcoding going on that you can't control - ie. outside your network, so I can go point to point from end-user phone to the people I peer with, but if they then transcode to G711 to go to the PSTN, it's OK, but if it then gets transcoded to GSM for a mobile, or back to G729 to go to an expensive overseas location, then quality does suffer )-: Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and G.726 Codec
Dear all, I am doing an interop testing with asterisk-1.6.0.5 now, and I have a question about the G.726 codec on asterisk. While my IAD supportes G.726-16,24,32 and 40 codecs, when doing a testing about G.726-40, I found that asterisk removed the G.72-40 sdp attrib when transmitting the INVITE with SDP. I modified sip.conf in order to solve the problem, G.726-32 is ok when allow=g726, but allow=g726-40 brings nothing. So I searched internet about the codec supporting in astersk, and found there is no certain words about it. following are some references: http://www.voip-info.org/wiki/index.php?page_id=127 said that Asterisk currently supports the 32kbps standard only, but didn't find anything about the asterisk version http://www.voiptutor.net/voip-info/wiki/view/Asterisk+codecs.html said that G.726 - 32kbps in Asterisk 1.0.3, 16/24/32/40kbps in CVS HEAD; flawed until Asterisk 1.4 which corrected the implementation and introduced g726aal2 for backwards compatibility with Asterisk 1.2 installations , from the article I find that 16/24/32/40 maybe already supported in astersik, but how to config it? My questions: 1. G.726 16/24/32/40 supported in asterisk-1.6.0.5? 2. If supporeted, how to configurate? Thanks! Le'an Liu 乐安 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
