Hi everyone (Mark, Jim). I am new to the list but thanks to both Mark and Jim, I have being using "asterisk" since summer 2001. I am just updating my version that was a year old. Yes, I know, I got busy with other things like paying bills so I don't have to sleep with the dog anymore. Anyway, one of the frustrations I have been dealing with (keep in mind that my version of asterisk (it was called -ng), libpri, zaptel and zapata are old) is that they leave the lines open (loopstart lines using the kwelstart in asterisk and zapata) when I receive a call from the PSTN and the asterisk PBX creates a bridge to connect to another line (PSTN) going out. No phones involved. I have the old Zapata/Tormenta ISA T1's (great job done by Jim) and I am using my unit as a PBX at home. I found that we were not ready for prime time yet so I have been waiting. The channel bank I use is the Atlas TA 750 with both FXO and FXS cards. I never had this problem when I had loopstart lines from the PSTN for incoming calls and trunks (groundstart) lines for terminating calls in the PSTN. I just placed an order for those lines at home and I am going to have to pay big time because they are putting a T1 into my house just for this purpose (and charge me the installation but not the monthly, I hope in time I can convert to a PRI-ISDN). I also use the Wildcards (both USB and PCI versions) in another unit I have in Colombia plus one in Jamaica. If that was not enough, I do have an anoying problem when I am on a VoIP call to another unit. When a call comes in from the PSTN and it is taken by, say my wife (the dog has not learned how to answer yet), I find that consistently the VoIP goes simplex (i.e. the other side (with asterisk and either wildcards or Tormenta) end up loosing reception. am I alone? Thank you guys. (Jim, by the way, if you are out there, I still got you defined in iax). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030407/1c1fe31b/attachment.htm
It is funny that you mention this- when I tried to make a PSTN bridged call through a E400 and a SIP phone this morning I had the exact same problem. I couldn't do a trace because I had to run out of the house early. Will trace next time it happens. -GSR -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Uriel Carrasquilla Sent: 08 April 2003 05:21 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] I must be alone Hi everyone (Mark, Jim). I am new to the list but thanks to both Mark and Jim, I have being using "asterisk" since summer 2001. I am just updating my version that was a year old. Yes, I know, I got busy with other things like paying bills so I don't have to sleep with the dog anymore. Anyway, one of the frustrations I have been dealing with (keep in mind that my version of asterisk (it was called -ng), libpri, zaptel and zapata are old) is that they leave the lines open (loopstart lines using the kwelstart in asterisk and zapata) when I receive a call from the PSTN and the asterisk PBX creates a bridge to connect to another line (PSTN) going out. No phones involved. I have the old Zapata/Tormenta ISA T1's (great job done by Jim) and I am using my unit as a PBX at home. I found that we were not ready for prime time yet so I have been waiting. The channel bank I use is the Atlas TA 750 with both FXO and FXS cards. I never had this problem when I had loopstart lines from the PSTN for incoming calls and trunks (groundstart) lines for terminating calls in the PSTN. I just placed an order for those lines at home and I am going to have to pay big time because they are putting a T1 into my house just for this purpose (and charge me the installation but not the monthly, I hope in time I can convert to a PRI-ISDN). I also use the Wildcards (both USB and PCI versions) in another unit I have in Colombia plus one in Jamaica. If that was not enough, I do have an anoying problem when I am on a VoIP call to another unit. When a call comes in from the PSTN and it is taken by, say my wife (the dog has not learned how to answer yet), I find that consistently the VoIP goes simplex (i.e. the other side (with asterisk and either wildcards or Tormenta) end up loosing reception. am I alone? Thank you guys. (Jim, by the way, if you are out there, I still got you defined in iax). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030408/938f1241/attachment.htm