Is there a limit to the number of parked calls Asterisk can handle?
Thanks,
Matt
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
Jonas Kellens wrote:
So if I understand correct, you don't need to look at the amount of
concurrent calls to calculate the RTP range in rtp.conf, you need to
look at the amount of INVITES that are being send at one moment ?
The number of concurrent channels in existence which are using RTP.
On 10/29/2013 05:14 PM, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
short question : does Asterisk reserve RTP ports for every IP-phone that
is being called ?
It uses 2 ports per channel under normal circumstances, 1 for RTP and
1 for RTCP.
If for instance an incoming call makes 10 IP-
Hello,
short question : does Asterisk reserve RTP ports for every IP-phone that
is being called ?
If for instance an incoming call makes 10 IP-phones ring, does this mean
that Asterisk preserves 10 x 2 RTP ports for audio ?
I guess Asterisk sends in the SIP INVITE an SDP body with an RTP po
Jonas Kellens wrote:
Hello,
short question : does Asterisk reserve RTP ports for every IP-phone that
is being called ?
It uses 2 ports per channel under normal circumstances, 1 for RTP and 1
for RTCP.
If for instance an incoming call makes 10 IP-phones ring, does this mean
that Asterisk pr
Hi
We have come across a situation where we are loosing synch of party 1 &
party 2 voice in call recording.
Here is the scenario
Party 1 initiate a call to Party 2 using AMI commands
When both calls are connected, we bridge these 2 calls. Then we start
recording of this bridged call using AMI
Hi there,
In other words you are maybe on 60ms and they are on 20ms or vice versa.
Do a wireshark trace and see if the codecs and ptime agree on both sides
otherwise you will get grabbled sounds.
On 10/29/2013 02:49 PM, Daniel van den Berg wrote:
> Hi there,
>
> Sounds like codec ptime mismatch..
Hi there,
Sounds like codec ptime mismatch...what codec are you using? If you are
using g729 make sure that you and your provider is giving the same ptime.
On 10/29/2013 11:55 AM, Stelios Koroneos wrote:
> On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote:
>> All,
>>
>>
>> The users in our or
On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote:
> All,
>
>
> The users in our organization are well, quite frankly, sick of phone
> service that is being provided. The choppy phone calls, and drop outs
> are detrimental to our sales force.
>
>
> I've tried about everything I can think o