Re: [asterisk-users] Codec choice
This is at least the third post under the subject 'Codec Choice' by the same sender. Why don't you stay within your first thread? Does posting over and over again increases chances of getting a solution? If so, then maybe I should try the same, as seems like an increasing trend on this list. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-24 7:13 AM, "Deepika Nijhawan" wrote: Hi, Group () and Group_Count () will need to be used on certain extension. What if there are lot of clients on the kit with different routings some going to dahdi and some to different sip interconnects, how can we do it on whole kit basis. Or let me know if there is any other way to use these functions to achieve this. Thanks, 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 -- _ -- 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] Codec choice
Hi, Group () and Group_Count () will need to be used on certain extension. What if there are lot of clients on the kit with different routings some going to dahdi and some to different sip interconnects, how can we do it on whole kit basis. Or let me know if there is any other way to use these functions to achieve this. Thanks, 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] Codec choice
On Fri, 20 Aug 2010, Sherwood McGowan wrote: > Good point my man...You drinking yet? Let me check to see if I still have a pulse -- yep! -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Codec choice
Steve, Good point my man...You drinking yet? LOL...I had forgotten about the GROUP and GROUP_COUNT functions, that is a much better way (in that it already existed and doesn't require me to write more code :] ) Slainte! On Fri, Aug 20, 2010 at 7:37 PM, Steve Edwards wrote: > On Fri, 20 Aug 2010, Sherwood McGowan wrote: > >> 1. Set up a Global Variable that will store that kit's current number of >> calls >> 2. Check that variable when a call starts (but before you dial out) >> 3. If the number of calls is <49 (since the current call will make >> 50), use codec A via the CHANNEL() function, otherwise use codec B >> using the same function. >> 4. Increment the variable >> 5. place call >> 6., upon hangup, decrement the variable > > Not really paying close attention to what you're trying to do, but... > > The GROUP() and GROUP_COUNT() functions automagically take care of the > increment and decrement cruft in a "race condition free" sort of way. > > Both methods still leave a small "window of opportunity" in comparing the > count with the threshold. > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- 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] Codec choice
On Fri, 20 Aug 2010, Sherwood McGowan wrote: > 1. Set up a Global Variable that will store that kit's current number of calls > 2. Check that variable when a call starts (but before you dial out) > 3. If the number of calls is <49 (since the current call will make > 50), use codec A via the CHANNEL() function, otherwise use codec B > using the same function. > 4. Increment the variable > 5. place call > 6., upon hangup, decrement the variable Not really paying close attention to what you're trying to do, but... The GROUP() and GROUP_COUNT() functions automagically take care of the increment and decrement cruft in a "race condition free" sort of way. Both methods still leave a small "window of opportunity" in comparing the count with the threshold. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Codec choice
1. Set up a Global Variable that will store that kit's current number of calls 2. Check that variable when a call starts (but before you dial out) 3. If the number of calls is <49 (since the current call will make 50), use codec A via the CHANNEL() function, otherwise use codec B using the same function. 4. Increment the variable 5. place call 6., upon hangup, decrement the variable Cheers On Fri, Aug 20, 2010 at 9:06 AM, Deepika Nijhawan wrote: > Hi, > > > > Thanks. Actually can it be done on whole kit basis rather than for an > extension or peer. Like if there are lot of inbound sip interconnects on a > kit , how can we send first 50% simultaneous calls to dahdi with codec A and > after that with codec B. > > > > Thanks, > > 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 > -- _ -- 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] Codec choice
Hi, Thanks. Actually can it be done on whole kit basis rather than for an extension or peer. Like if there are lot of inbound sip interconnects on a kit , how can we send first 50% simultaneous calls to dahdi with codec A and after that with codec B. Thanks, 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] Codec choice
> On Thu, Aug 19, 2010 at 3:14 AM, Deepika Nijhawan >> >> Does anyone has an idea how to tell asterisk to use codec A for first >> 50 calls and then codec B for rest of the calls. On Thu, 19 Aug 2010, Sherwood McGowan wrote: > the easiest way I can think of is to use a global variable that you > increment each time a new call spawns, and once it's over your threshold > (50 in this case) use the CHANNEL() function to set the audio format to > the codec you want (google voip-info function CHANNEL) Your question is not specific enough. Do you mean the "first 50 calls" or "50 simultaneous calls?" I suspect the latter and the GROUP() and GROUP_COUNT() functions are the way to go. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Codec choice
Ok. And how will we do for getting sip inbound calls from different ips and sending them to dahdi. Thanks, 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] Codec choice
i do this by having 2 peers setup, one has a call limit of 10 and uses g729, the rest of the calls get sent to the second peer which uses ulaw. all calls attempt peer 1 if there's channels available it uses it if not it just moves through the dialplan to use the second one. On 19 August 2010 09:14, Deepika Nijhawan wrote: > Hi, > > > > Does anyone has an idea how to tell asterisk to use codec A for first 50 > calls and then codec B for rest of the calls. > > > > Thanks, > > Deepika > > -- > _ > -- 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] Codec choice
On Thu, Aug 19, 2010 at 3:37 AM, Steve Howes wrote: > > On 19 Aug 2010, at 09:14, Deepika Nijhawan wrote: >> Does anyone has an idea how to tell asterisk to use codec A for first 50 >> calls and then codec B for rest of the calls. > > You could create two separate trunks, one for each codec? > > S > -- > _ > -- 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 > Good Point! LOL, I went with a much more complicated method...sleep deprivation at it's finest perhaps? -- _ -- 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] Codec choice
On 19 Aug 2010, at 09:14, Deepika Nijhawan wrote: > Does anyone has an idea how to tell asterisk to use codec A for first 50 > calls and then codec B for rest of the calls. You could create two separate trunks, one for each codec? S -- _ -- 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] Codec choice
On Thu, Aug 19, 2010 at 3:14 AM, Deepika Nijhawan wrote: > Hi, > > > > Does anyone has an idea how to tell asterisk to use codec A for first 50 > calls and then codec B for rest of the calls. > > > > Thanks, > > Deepika > > -- > _ > -- 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 > the easiest way I can think of is to use a global variable that you increment each time a new call spawns, and once it's over your threshold (50 in this case) use the CHANNEL() function to set the audio format to the codec you want (google voip-info function CHANNEL) Cheers -- _ -- 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] Codec choice
Hi, Does anyone has an idea how to tell asterisk to use codec A for first 50 calls and then codec B for rest of the calls. Thanks, Deepika -- _ -- 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
