RE: [Asterisk-Users] Zap call bridge drops randomly
Don't know for sure, but it happens on the TDM card presumably due to asterisk translating certain voices into dtmf. Busycount=6 corrects the problem on it; presumably it would on a T1 as well. Would enabling Busydetect really help if Asterisk thinks it detects an On-Hook? MATT--- -Original Message- We have a call redirection system setup inhouse to send calls from an incoming line on a T1 to an external dialed out number: Zap(call comes in) - Asterisk - Zap(call dials out) The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a PRI. We are using Asterisk release 1.0.5 Randomly the calls will drop less than a minute into the call. The Debug messages at the end of the call always say something like this: (incoming call on 58, outgoing on 73) Feb 21 11:37:28 DEBUG[16804]: Exception on 63, channel 58 Feb 21 11:37:28 DEBUG[16804]: Got event On hook(1) on channel 58 (index 0) Feb 21 11:37:28 DEBUG[16804]: disabled echo cancellation on channel 58 Feb 21 11:37:28 DEBUG[16804]: Unlinking slave 73 from 58 The strange thing is that the person that called in did not hang up, in fact they are usually talking when this call goes dead. Only about 10% of the calls that redirect have this happen to them, and it seems to be random as to which ones drop. Calls that dial out on either T1 and calls that come in and are not redirected never seem to have these problems. I have callprogress=no and busydetect=no but that doesn't seem to help. Anyone have an idea on this? Is there any way to make Asterisk less sensitive to hangups if that's even the cause? Just looking for some feedback before I post on the bugtracker. You might try: busydetect=yes busycount=6 in the top section of zapata.conf and see if that helps. I've not tried this on a T1, but it certainly clears up the same issue on the TDM pstn lnterfaces. ___ 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] Zap call bridge drops randomly
We have a call redirection system setup inhouse to send calls from an incoming line on a T1 to an external dialed out number: Zap(call comes in) - Asterisk - Zap(call dials out) The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a PRI. We are using Asterisk release 1.0.5 Randomly the calls will drop less than a minute into the call. The Debug messages at the end of the call always say something like this: (incoming call on 58, outgoing on 73) Feb 21 11:37:28 DEBUG[16804]: Exception on 63, channel 58 Feb 21 11:37:28 DEBUG[16804]: Got event On hook(1) on channel 58 (index 0) Feb 21 11:37:28 DEBUG[16804]: disabled echo cancellation on channel 58 Feb 21 11:37:28 DEBUG[16804]: Unlinking slave 73 from 58 The strange thing is that the person that called in did not hang up, in fact they are usually talking when this call goes dead. Only about 10% of the calls that redirect have this happen to them, and it seems to be random as to which ones drop. Calls that dial out on either T1 and calls that come in and are not redirected never seem to have these problems. I have callprogress=no and busydetect=no but that doesn't seem to help. Anyone have an idea on this? Is there any way to make Asterisk less sensitive to hangups if that's even the cause? Just looking for some feedback before I post on the bugtracker. You might try: busydetect=yes busycount=6 in the top section of zapata.conf and see if that helps. I've not tried this on a T1, but it certainly clears up the same issue on the TDM pstn lnterfaces. ___ 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] Zap call bridge drops randomly
Would enabling Busydetect really help if Asterisk thinks it detects an On-Hook? MATT--- -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Monday, February 21, 2005 7:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zap call bridge drops randomly We have a call redirection system setup inhouse to send calls from an incoming line on a T1 to an external dialed out number: Zap(call comes in) - Asterisk - Zap(call dials out) The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a PRI. We are using Asterisk release 1.0.5 Randomly the calls will drop less than a minute into the call. The Debug messages at the end of the call always say something like this: (incoming call on 58, outgoing on 73) Feb 21 11:37:28 DEBUG[16804]: Exception on 63, channel 58 Feb 21 11:37:28 DEBUG[16804]: Got event On hook(1) on channel 58 (index 0) Feb 21 11:37:28 DEBUG[16804]: disabled echo cancellation on channel 58 Feb 21 11:37:28 DEBUG[16804]: Unlinking slave 73 from 58 The strange thing is that the person that called in did not hang up, in fact they are usually talking when this call goes dead. Only about 10% of the calls that redirect have this happen to them, and it seems to be random as to which ones drop. Calls that dial out on either T1 and calls that come in and are not redirected never seem to have these problems. I have callprogress=no and busydetect=no but that doesn't seem to help. Anyone have an idea on this? Is there any way to make Asterisk less sensitive to hangups if that's even the cause? Just looking for some feedback before I post on the bugtracker. You might try: busydetect=yes busycount=6 in the top section of zapata.conf and see if that helps. I've not tried this on a T1, but it certainly clears up the same issue on the TDM pstn lnterfaces. ___ 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
