Re: [Asterisk-Users] CAS SF Inband tone signalling problem
Channel Associated Signaling is a simple 4 bits 'protocol' often used in digital non-isdn E1 connections to define/indicate the state of the line (or channel): sieze, sieze ack, idle, etc, at a given moment. Usualy, only 2 of those 4 bits are used. However, given that CAS tables may vary from manufacturer to manufacturer, the most important thing to start with in a PBX-to-PBX CAS connection is to identify the CAS table being used at the far-end switch. If you can not easily obtain the tables on the far-end --which is the usually the case when the brand of your switch is not the same as the one at the far end, or if you don't have previous experience with CAS, you may get stuck even if you have a CAS analyzer. Given that in PBX-to-PBX scenarios you usually go with E&M signaling, you will also have to deal with the start arrangement; usually WiNK or IMMediate You have to have the same start arrangemnet for each direction (REC/XMT) as the far end switch. If Wink Start, you also have to deal with the length of the wink pulse. If you have the far end CAS table, just program yours to match their values. I am new with Asterisk and have never used zaptel.conf, but if u attach the far end CAS table to your post, we can be more precise in helping. Rgds Sam\\\ - Original Message - From: M.A. Ali To: [EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 1:05 AM Subject: [Asterisk-Users] CAS SF Inband tone signalling problem Hi, I am having some problem in defining CAS SF Inband signalling on a Digium E100P card. The problem is that the syntax given in the sample zaptel.conf doesn't work. Can someone provide me with an example of the syntax. I am trying to connnect asterisk with another exchange using a E1 CAS. I guess SF is the only option as the rest fxo,fxs,e&m etc are all subscriber end signalling. i need a CO to CO (or PBX to PBX) signalling...Somebody guide me on that too. Any help will be highly appreciated. Thnaks Sincerly, Janjua The new MSN 8: smart spam protection and 2 months FREE* ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] CAS SF Inband tone signalling problem
E&M is the traditional protocol to connect two COs or PBXs together. E&M trunks are inherently symmetrical. So there is no subscriber or CO side. You have to see about the higher level signalling protocol to make sure dialed digits etc work right. Regards. Alfred R. Nurnberger F L O S Y SMaking Communications FlowTel: +1 (503) 972-9300Fax: +1 (503) 972-9309US Toll Free: 1-877-4FLOSYShttp://www.flosys.com -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of M.A. AliSent: Tuesday, January 20, 2004 11:06 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] CAS SF Inband tone signalling problem Hi, I am having some problem in defining CAS SF Inband signalling on a Digium E100P card. The problem is that the syntax given in the sample zaptel.conf doesn't work. Can someone provide me with an example of the syntax. I am trying to connnect asterisk with another exchange using a E1 CAS. I guess SF is the only option as the rest fxo,fxs,e&m etc are all subscriber end signalling. i need a CO to CO (or PBX to PBX) signalling...Somebody guide me on that too. Any help will be highly appreciated. Thnaks Sincerly, Janjua The new MSN 8: smart spam protection and 2 months FREE* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAS SF Inband tone signalling problem
Hi, I am having some problem in defining CAS SF Inband signalling on a Digium E100P card. The problem is that the syntax given in the sample zaptel.conf doesn't work. Can someone provide me with an example of the syntax. I am trying to connnect asterisk with another exchange using a E1 CAS. I guess SF is the only option as the rest fxo,fxs,e&m etc are all subscriber end signalling. i need a CO to CO (or PBX to PBX) signalling...Somebody guide me on that too. Any help will be highly appreciated. Thnaks Sincerly, Janjua The new MSN 8: smart spam protection and 2 months FREE* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users