[Asterisk-Users] Choppy Sound on PSTN End
Title: Choppy Sound on PSTN End Hi all, I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux. Our call routing is like this: SJPHONE on Windows - QoS-enabled Switch - Asterisk - T1 Line - Broadvoice SIP account - PSTN Calls seem to work great from user to user. However, calls from a SJPhone user to the PSTN are not so great. The SJPhone user hears the person on the PSTN perfectly I mean, completely flawless. However, the user on the PSTN end hears choppy / jittery, extraneous clicks, etc. Here is the SJPhone config: Audio Compression: G.711 Driver buffer size: 20 msec Driver input queue length: 6 Driver output queue length: 4 RTP jitter queue length: 6 Silence Suppression: No DTMF Sending: RFC 2833 Signal Duration (ms): 270 RTP Payload type: 101 Signal volume: 10 Pause duration (ms): 100 And the sip extension config (in Asterisk Management Portal): Allow: blank Canreinvite: no Disallow: gsm Dtmfmode: rfc2833 Host: dynamic Nat: yes (some users are behind NAT) Qualify: no Any ideas on what to do to get rid of the choppiness? Thanks! Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Choppy Sound on PSTN End
I doubt it is the RAID controller since my Dell server isn't using one and I have this problem... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aza Sent: Monday, May 02, 2005 11:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Choppy Sound on PSTN End I have the same problem on a Dell 1850 with a TE410P and have been attempting to narrow it down. Interrupts don't seem to be a problem and I have two PRIs from two different suppliers and both have the same static/chop on the line so it's not the PRI. The leading suspect at the moment is the RAID controller. Unfortunately it's rather difficult to remove this from the set up but I plan to switch one of the PRIs to a Dell 1750 without a RAID controller to see if the problem still goes away. Aaron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Choppy Sound on PSTN End
Title: Choppy Sound on PSTN End I turned on qualify=yes to see what happens. No effect that I can see. I have GSM disabled because I heard some bad things about the GSM protocol. I reenabled it, but to no avail. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Monday, May 02, 2005 11:22 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Choppy Sound on PSTN End I have the exact setup you describe, SJPhone - * - Zap/PRI. I think you need to twiddle some settings. You might turn on qualify just to see if the * is seeing network flaws. Keep in mind, if your using windows, anytime the user starts clicking around, you can expect less than ideal audio. Also, why disable GSM ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI Question
Let me further clarify this. I am looking to buy the TE110P. The website says that The TE110P with Asterisk will route voice and data traffic, and eliminate the need for an external router. How does this work? How is the data transferred - as a pass-through like a NAT to the server's network card? What kind of network slowdown are we looking at? How does this affect the processor? I would appreciate some more information on how this works. Thanks, Tim Chandler -- Original Message -- From: Tim Chandler [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 21 Mar 2005 21:07:39 -0700 Hi Everyone, Thanks for all the input you add to the list. This seems to be a very good list. I am still new to Asterisk. If I run a PRI integrated T1 line into my office, do I need to split the line between the data and voice before plugging it into the asterisk box or is there some other way to do that? What are some good options for splitting the line? Thanks for any input. Tim BTW - Giving everyone a hug is an expression in Brazil. Everyone says it... it's like saying have a good one or good to see you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Choosing an ISP for Asterisk
I am the IT Manager for an international company who preserves its competitive edge by cutting costs. We are moving to a new office in about two months, and naturally, Asterisk came to mind as a way to implement a VoIP setup at low cost. My expertise is computers, not telephones, so all of this is new to me. I need to know what the ideal setup for an Asterisk set up is. My idea is to have our ISP run a voice/data T1 into our corporate office (about 60 users) and run this into a fast Dell server with Asterisk and a Digium card. Theoretically, the ISP would connect us into the traditional phone system. However, I don't know the specific requirements for the T1 line or how to split the data and voice. What is the best type of T1 line to run? What protocols does it need to support? I have heard of dynamic T1s, how do you split the data and voice? We are thinking of using someone like XO (who currently does both our voice and data). What do I need for this installation? I appreciate your help in this! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
