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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/6114bf1d/attachment.htm
On Thu, Aug 19, 2010 at 3:14 AM, Deepika Nijhawan <deepika.nijhawan at oxygen8.com> 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
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
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 <deepika.nijhawan at oxygen8.com>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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/d6833d57/attachment.htm
Ok. And how will we do for getting sip inbound calls from different ips and
sending them to dahdi.
Thanks,
D
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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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100820/c8a9c913/attachment.htm
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 <deepika.nijhawan at oxygen8.com> 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 >
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100824/e319a000/attachment.htm
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" <deepika.nijhawan at
oxygen8.com>
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
--
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