RE: [Asterisk-Users] Choppy Sound on PSTN End
That's good to know about the RAID controller not being the problem, will save me some testing. I wonder if it's a change in asterisk then? Previously I was using E100Ps with asterisk-1.0.0 and didn't have the chop. Unfortunately at the same time I put in the TE410P I upgraded to asterisk-1.0.7 so don't know if the problem was introduced by the move from E100P to TE410P or from 1.0.0 to 1.0.7. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Chandler Sent: 02 May 2005 18:32 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: 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 ___ 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] 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
Hi, I have the same problem on a Dell 1850 with a TE410P, static/chop on calls to through the TE410P, and have been attempting to narrow it down for the last week. 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 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Chandler Sent: 02 May 2005 17:23 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 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
Title: 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 ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim ChandlerSent: Monday, May 02, 2005 11:23 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] 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 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 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Chandler Sent: 02 May 2005 17:23 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 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
[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
