Re: [asterisk-users] One Way Delay in Audio Over Analog
Thanks for the replies! The analog card is a TDM400P. The system is currently using asterisk-1.4 r81383 and zaptel-1.4 r2649. I am told that the problem did not exist when the system was using asterisk-1.2.10 and zaptel 1.2.9.1. -Brian On 12/31/07, Brian Alexander <[EMAIL PROTECTED]> wrote: > > I have been trying to track down the cause/fix for a problem and I am out > of ideas... I am hoping one of you can point me in the right direction. > > The symptom is that when a calls is placed from an internal extension > through an analog line to a number on the pstn the caller can hear the > callee but the callee can not hear the caller for as long as ten seconds. > > The problem appears to happen fairly consistently on the same pstn > numbers. However, I have not seen a common characteristic in those numbers. > For example, one of them is a direct number to a cell phone and another is > to a Verizon fiber-optic phone/data service. > > The problem does not seem to be related to the type of SIP phone being > used by the caller - for example, we have tried both X-Lite and Polycom > phones without a change in behavior. > > The problem does not appear to occur if the callee then calls into our > system (at least the one time I was able to have this happen). > > Turning on or off echo cancellation and/or call progress does not seem to > change the behavior. > > I will appreciate any ideas you have. I am certainly stumped. > > Thanks and Happy New Year! > -Brian > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Delay in Audio Over Analog
What are you using for a PSTN gateway? From: Brian Alexander [mailto:[EMAIL PROTECTED] Sent: Monday, December 31, 2007 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] One Way Delay in Audio Over Analog I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction. The symptom is that when a calls is placed from an internal extension through an analog line to a number on the pstn the caller can hear the callee but the callee can not hear the caller for as long as ten seconds. The problem appears to happen fairly consistently on the same pstn numbers. However, I have not seen a common characteristic in those numbers. For example, one of them is a direct number to a cell phone and another is to a Verizon fiber-optic phone/data service. The problem does not seem to be related to the type of SIP phone being used by the caller - for example, we have tried both X-Lite and Polycom phones without a change in behavior. The problem does not appear to occur if the callee then calls into our system (at least the one time I was able to have this happen). Turning on or off echo cancellation and/or call progress does not seem to change the behavior. I will appreciate any ideas you have. I am certainly stumped. Thanks and Happy New Year! -Brian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Delay in Audio Over Analog
Brian Alexander wrote: > I have been trying to track down the cause/fix for a problem and I am > out of ideas... I am hoping one of you can point me in the right direction. > > The symptom is that when a calls is placed from an internal extension > through an analog line to a number on the pstn the caller can hear the > callee but the callee can not hear the caller for as long as ten seconds. > > The problem appears to happen fairly consistently on the same pstn > numbers. However, I have not seen a common characteristic in those > numbers. For example, one of them is a direct number to a cell phone and > another is to a Verizon fiber-optic phone/data service. > > The problem does not seem to be related to the type of SIP phone being > used by the caller - for example, we have tried both X-Lite and Polycom > phones without a change in behavior. > > The problem does not appear to occur if the callee then calls into our > system (at least the one time I was able to have this happen). > > Turning on or off echo cancellation and/or call progress does not seem > to change the behavior. > > I will appreciate any ideas you have. I am certainly stumped. > > Thanks and Happy New Year! > -Brian Brian, What about some facts ? Hardware ? Software versions ? /Mats ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One Way Delay in Audio Over Analog
I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction. The symptom is that when a calls is placed from an internal extension through an analog line to a number on the pstn the caller can hear the callee but the callee can not hear the caller for as long as ten seconds. The problem appears to happen fairly consistently on the same pstn numbers. However, I have not seen a common characteristic in those numbers. For example, one of them is a direct number to a cell phone and another is to a Verizon fiber-optic phone/data service. The problem does not seem to be related to the type of SIP phone being used by the caller - for example, we have tried both X-Lite and Polycom phones without a change in behavior. The problem does not appear to occur if the callee then calls into our system (at least the one time I was able to have this happen). Turning on or off echo cancellation and/or call progress does not seem to change the behavior. I will appreciate any ideas you have. I am certainly stumped. Thanks and Happy New Year! -Brian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
