Re: [asterisk-users] 3rd party app store
On 09/19/2010 12:06 AM, Darren Nickerson wrote: > On Sep 18, 2010, at 11:41 AM, Mark Deneen wrote: > >> On Fri, Sep 17, 2010 at 11:52 PM, Dean Collins wrote: >>> Any thoughts on why the lack of traffic? >>> >>> >>> Cheers, >>> Dean >> >> Not enough applications to play immature bathroom sounds. > You could well be right, but consider for a moment a few alternatives. > > Perhaps it's the $5000 up front just to be listed? I see the fee's reduced to > $2500 now as a promo, but still that's a huge barrier for most. Even $1 will keep most free solutions out of a forum like that, so a blanket fee strategy must have been specifically chosen to skew things in a particular way. Seems like it worked very well. > Or perhaps its the fact that the nature of the apps that get listed means > they aren't usually 'purchase-able' with a simple 'click to buy' (how do you > sell SIP trunking with a click-to-buy???) - and as a consequence there's no > purchase capability built into the asteriskexchange site, just link outs to > different purchase-ish URLs for the various products. Anyone looking to sell > their app would need to develop their own point-of-sale/payment processing > systems so it's really not an 'app store' at all in the traditional > sense. That is a pretty basic problem for some things, but not for everything. Plenty of telephony stuff is a "thing" for sale, even if some after sales support is needed, to get over installation issues. > Kudos to digium for realizing this goal, but I think the $5000 get-in cost > has resulted in the lack of interest/popularity, and limited the listings to > only the largest, most profitable asterisk/digium partners. > Kudos to Digium for taking an idea that could have worked against their interests, and sidelining it so well nobody created a real marketplace. The bottom line, of course, is that if people like regular posters here didn't know about about the site, the real target audience most certainly does not. Nothing more is needed to explain the low traffic. Even if you are serious about creating a vibrant, orderly marketplace, its really hard. Look at the variation in quality between them. Even Google, which is basically a marketing company, seem to have no idea how to make the Android market function. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A108 PCIe V2.0
Those modifications are done via regular Sangoma installation with a special option to the Setup script. http://wiki.sangoma.com/wanpipe-linux-asterisk-appendix#zaptel_adjustable_chunk_sz http://www.sangoma.com/assets/docs/misc/2009_10_09_How_to_Reduce_Asterisk_System_Loads.pdf Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com On Sat, Sep 18, 2010 at 9:45 AM, Nyamul Hassan wrote: > Thank you for your info Moises. For those who want to have a high density > system, can you provide what modifications to the Dahdi (or anything else) > do you make? > > Regards > HASSAN > > > On Sat, Sep 18, 2010 at 19:39, Moises Silva wrote: > >> >> On Fri, Sep 17, 2010 at 11:22 AM, Nyamul Hassan wrote: >> >>> While this is too many "eggs" in one basket, but can be useful if you >>> have "too many" E(T)1s say equivalent to a STM1 (OC3) or more. In that >>> case, it would be too many boxes at 8ports / box. >>> >>> Somewhere in the mailing list, Sangoma devs said that they do 32E(T)1 >>> per box on the labs quite frequently, although mostly for load testing. >>> >>> >> That is correct, that is our typical load test scenario. However, Asterisk >> is a complex system with many features. Our testing focus on SIP to TDM >> bridging, meaning the only used applications are Answer() and Dial() with >> the DAHDI and SIP channel drivers, typically with latest 1.4. >> >> Additionally we always compile DAHDI modifying the chunk size to reduce >> the interrupt load. >> >> As far as your question about PCIe 2.0, yes the A108 should work just fine >> there. >> >> Moises Silva >> Senior Software Engineer >> Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON >> L3R 9R6 Canada >> t. 1 905 474 1990 x128 | e. m...@sangoma.com >> >> >> > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] externip/localnet
On Sat, 18 Sep 2010 00:54:38 +0100, dotnetdub wrote > Hi All, > > Is it possible to specify more than 1 localnet? I know this is an odd > question. I have a customer that has multiple sites linked by VPN. > > Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24 > > We want to allow some access to the public IP address at the main site. For > this to work I need to use the externip and localnet directive. If I do this > it rewrites the SDP with the external IP address of the main site on dialog > with the VPN'd sites. > > This means that I can either have the VPN endpoints working or have people > accessing from outside.. > You can put as many localnet statements as you need, one per line. I have similar setups where up to 5 networks are internally connected so I just do something like: localnet=192.168.11.0/24 localnet=192.168.16.0/24 ... -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd party app store
On Sep 18, 2010, at 11:41 AM, Mark Deneen wrote: > On Fri, Sep 17, 2010 at 11:52 PM, Dean Collins wrote: >> Any thoughts on why the lack of traffic? >> >> >> Cheers, >> Dean > > > Not enough applications to play immature bathroom sounds. You could well be right, but consider for a moment a few alternatives. Perhaps it's the $5000 up front just to be listed? I see the fee's reduced to $2500 now as a promo, but still that's a huge barrier for most. Or perhaps its the fact that the nature of the apps that get listed means they aren't usually 'purchase-able' with a simple 'click to buy' (how do you sell SIP trunking with a click-to-buy???) - and as a consequence there's no purchase capability built into the asteriskexchange site, just link outs to different purchase-ish URLs for the various products. Anyone looking to sell their app would need to develop their own point-of-sale/payment processing systems so it's really not an 'app store' at all in the traditional sense. Kudos to digium for realizing this goal, but I think the $5000 get-in cost has resulted in the lack of interest/popularity, and limited the listings to only the largest, most profitable asterisk/digium partners. -d -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?
Sorry i don't really understand your message ;=) my english are bad. I am search a sample of configuration of the audiocode. 2010/9/18 Paul Belanger : > On Sat, Sep 18, 2010 at 6:46 AM, Olivier CALVANO wrote: >> Anyone use this equipements with asterisk ? because i am search a >> config sample for AudioCode and for Asterisk (i am new in VoIP). >> > Why would you want too? Asterisk can do everything, and more, then > the Audiocodes. > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd party app store
On Fri, Sep 17, 2010 at 11:52 PM, Dean Collins wrote: > Any thoughts on why the lack of traffic? > > > Cheers, > Dean Not enough applications to play immature bathroom sounds. Just a guess. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determine busy state
On Sat, Sep 18, 2010 at 6:54 AM, wrote: > What am I doing wrong? > Enable a SIP debug and find out. http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?
On Sat, Sep 18, 2010 at 6:46 AM, Olivier CALVANO wrote: > Anyone use this equipements with asterisk ? because i am search a > config sample for AudioCode and for Asterisk (i am new in VoIP). > Why would you want too? Asterisk can do everything, and more, then the Audiocodes. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A108 PCIe V2.0
Thank you for your info Moises. For those who want to have a high density system, can you provide what modifications to the Dahdi (or anything else) do you make? Regards HASSAN On Sat, Sep 18, 2010 at 19:39, Moises Silva wrote: > > On Fri, Sep 17, 2010 at 11:22 AM, Nyamul Hassan wrote: > >> While this is too many "eggs" in one basket, but can be useful if you have >> "too many" E(T)1s say equivalent to a STM1 (OC3) or more. In that case, it >> would be too many boxes at 8ports / box. >> >> Somewhere in the mailing list, Sangoma devs said that they do 32E(T)1 >> per box on the labs quite frequently, although mostly for load testing. >> >> > That is correct, that is our typical load test scenario. However, Asterisk > is a complex system with many features. Our testing focus on SIP to TDM > bridging, meaning the only used applications are Answer() and Dial() with > the DAHDI and SIP channel drivers, typically with latest 1.4. > > Additionally we always compile DAHDI modifying the chunk size to reduce the > interrupt load. > > As far as your question about PCIe 2.0, yes the A108 should work just fine > there. > > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON > L3R 9R6 Canada > t. 1 905 474 1990 x128 | e. m...@sangoma.com > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration attempts
I had to make a few minor edit in fail2ban to get it to work. I had to change the logger messages format for asterisk to: [general] dateformat=%F %T Thanks, Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Friday, September 17, 2010 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Registration attempts It means that fail2ban is not configured correctly on your machine. For me it works fine, and in fact lately these registration/hack attempts have gone up significantly, thanks to cloud computing I guess. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-17 5:28 PM, "dave george" wrote: I am getting several hundred registration attempts on my aserterisk per minute. I have fail2ban installed but it's not stopping the attempts. Any suggestions. Whatever they are using is changing the userid on each attempt. Latest IP: 209.172.57.219 Thanks, Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A108 PCIe V2.0
On Fri, Sep 17, 2010 at 11:22 AM, Nyamul Hassan wrote: > While this is too many "eggs" in one basket, but can be useful if you have > "too many" E(T)1s say equivalent to a STM1 (OC3) or more. In that case, it > would be too many boxes at 8ports / box. > > Somewhere in the mailing list, Sangoma devs said that they do 32E(T)1 per > box on the labs quite frequently, although mostly for load testing. > > That is correct, that is our typical load test scenario. However, Asterisk is a complex system with many features. Our testing focus on SIP to TDM bridging, meaning the only used applications are Answer() and Dial() with the DAHDI and SIP channel drivers, typically with latest 1.4. Additionally we always compile DAHDI modifying the chunk size to reduce the interrupt load. As far as your question about PCIe 2.0, yes the A108 should work just fine there. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determine busy state
Hi all, to be able to transfer calls I have set call-limit to 2 for all of my peers. Now how can I determine if a peer is in busy state using the first line if I don't want to route a second call to it? Thanks in advance, Oliver -- What I found is when I use sip.conf instead of realtime and set call-limit to 2 and busylevel to 1 it works as I expected. As soon as a peer has the first call (line one busy) and I try to call this peer I get user busy. But using realtime with the same settings (call-limit 2 and busylevel 1) this does not work. The second call is etablished via the second line. What am I doing wrong? Oliver -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?
Hi i have buy a Audiocode Median 2000 VoIP Gateway and connect it on : 1 E1 30 channels 1 Lan Port Anyone use this equipements with asterisk ? because i am search a config sample for AudioCode and for Asterisk (i am new in VoIP). I want that all calls arrives on the AudioCode are sent to the asterisk by SIP (trunk ?) and all outgoing call from Asterisk are sent to the AudioCode. I don't want specify numbers on the audiocode, a +33* => Asterisk. Thanks for your help Olivier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
