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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050502/874085b3/attachment.htm
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: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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
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: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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
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: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tim Chandler Sent: Monday, May 02, 2005 11:23 AM 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050502/84153d61/attachment.htm