Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

2013-10-29 Thread Ron Wheeler
On 28/10/2013 4:12 PM, Mark Wiater wrote: On 10/28/2013 3:59 PM, Ron Wheeler said: I am reaching the same level of frustration. I have tried to find the source of the problems. We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk - No analogue. I don't have any problems

Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

2013-10-29 Thread Stelios Koroneos
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 of.

Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

2013-10-29 Thread Daniel van den Berg
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

Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

2013-10-29 Thread Daniel van den Berg
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

[asterisk-users] Loosing synch between party 1 party 2 voice in monitor recording

2013-10-29 Thread Amit Patkar | ATPL
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

Re: [asterisk-users] Question about how Asterisk works with RTP ports

2013-10-29 Thread Joshua Colp
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

[asterisk-users] Question about how Asterisk works with RTP ports

2013-10-29 Thread Jonas Kellens
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

Re: [asterisk-users] Question about how Asterisk works with RTP ports

2013-10-29 Thread Jonas Kellens
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

Re: [asterisk-users] Question about how Asterisk works with RTP ports

2013-10-29 Thread Joshua Colp
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.

[asterisk-users] No of parked calls limit

2013-10-29 Thread Matt Hamilton
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