Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?
Quoting Jonathan Moore [EMAIL PROTECTED]: On Tue, 13 Jan 2004, James Sharp wrote: I have a little more info on this. Following the suggestion of another post on this topic I tracked down an analog phone with lighted buttons powered by the phone connection. I directly connected the phone to one of my inbound lines and called it with my cell phone. Picked up the analog phone, verified call completion and then hung up my cell. I watched and waited for the lights to go out. Sure enough they did, but it took 8 seconds from the time of the hangup. After the flash more phone started emitting a dialtone sound. Is this correct? I was under the impression the voltage drop would happen almost immediately. Do you have another analog line that's on the same Central office as the line in question? The delay could be lag time betwee the time you hang up your cell phone, the cell provider MTSO processes the hang up, passes it on to their termination provider, who then passes it on to your termination provider, who then passes it on to you. I will try that, but the results were consistent with what we are seeing across the board. I will call from my pbx to my test line in the morning and time the results. I tried the test again today but using an analog line to call the test line (plexar to plexar) and it was in the same range. I think my particular test call was actually 11 seconds. So it doesn't seem to matter whether it is a cell call or a land line call there is a significant delay before the voltage drops. I also met with our SBC rep today and he brought a technical person with him. The techie seemed to think the delay was longer than it should be. They are going to check into it. They are also going to check to see if the delay for ground start lines is lower if nothing can be done to speed up the loop start signaling. Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
On Tue, 13 Jan 2004, Jonathan Moore wrote: LSRB = Loop Start with Reverse Battery I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. Try LSRB - it may work. -- Joel ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to a local phone contractor and SBC directly and no one seems to know what I am talking about. The phone contractor knew about the issue with other phone systems in the area but didn't know there was a way to fix it and SBC reps seem to never have heard of disconnect sup or calling party disconnect. I've never seen a line from SBC that DIDN'T come with disconnect supervision (some SBC line monkeys I know call it battery drop disconnect). The * Handbook refers to loop start with call sup as kewlstart are there other names for this protocol? One of the local contractors thought that SBC automatically drops line voltage on remote hangup, in which case I need to know what signalling to program into the ADIT 600's fxo channels. I also have the option of going to groundstart signalling if this would fix the problem, but it would cause some line downtime so it is not my preferred method. Kewlstart is also an alias for battery drop disconnect. The Adit 600 manual lists the following options for mapping FXO ports to the T1 DSO. DPT = Dial Pulse Termination EMDW = EM Delayed Wink start EMI = EM Immediate start EMICPD = EM Immediate Start with Calling Party Disconnect EMW = EM Wink start GS = Ground Start GSRB = Ground Start with Reverse Battery LS = Loop Start LSCPD = Loop Start Calling Party Disconnect LSRB = Loop Start with Reverse Battery VoIP = Voice over IP (CMG only) I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. That should be right. If you're really interested in looking, take a cheap voltmeter and put it across the line. If everyone is on hook, you'll see 48V. If someone goes off hook, you'll see it drop to about 6V. If you see a quick drop to 0V when the far end hangs up, you've got battery drop disconnect. I don't know if this has any relevance but I am also originating the clock source from the * side with Wildcard T1 card. That's really the only way it'll work. The channel bank can't generate clocking. ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
I have a little more info on this. Following the suggestion of another post on this topic I tracked down an analog phone with lighted buttons powered by the phone connection. I directly connected the phone to one of my inbound lines and called it with my cell phone. Picked up the analog phone, verified call completion and then hung up my cell. I watched and waited for the lights to go out. Sure enough they did, but it took 8 seconds from the time of the hangup. After the flash more phone started emitting a dialtone sound. Is this correct? I was under the impression the voltage drop would happen almost immediately. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Joel Maslak [EMAIL PROTECTED]: On Tue, 13 Jan 2004, Jonathan Moore wrote: LSRB = Loop Start with Reverse Battery I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. Try LSRB - it may work. -- Joel ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
If you don't have a voltmeter to look at this, try just listening on the line (using an analog telephone) when the far end hangs up. You should hear a distinct click-click on the line a second or two after they hang up. If you hear this, it's likely you are getting the required disconnect supervision from the telco. Note that many (most?) smaller private PBX's do not drop loop current on an analog line when the far end disconnects - but central office class switches usually do. It's not very scientific, but once you've heard one you can recognise it. regards Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sharp Sent: Tuesday, January 13, 2004 3:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600? Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to a local phone contractor and SBC directly and no one seems to know what I am talking about. The phone contractor knew about the issue with other phone systems in the area but didn't know there was a way to fix it and SBC reps seem to never have heard of disconnect sup or calling party disconnect. I've never seen a line from SBC that DIDN'T come with disconnect supervision (some SBC line monkeys I know call it battery drop disconnect). The * Handbook refers to loop start with call sup as kewlstart are there other names for this protocol? One of the local contractors thought that SBC automatically drops line voltage on remote hangup, in which case I need to know what signalling to program into the ADIT 600's fxo channels. I also have the option of going to groundstart signalling if this would fix the problem, but it would cause some line downtime so it is not my preferred method. Kewlstart is also an alias for battery drop disconnect. The Adit 600 manual lists the following options for mapping FXO ports to the T1 DSO. DPT = Dial Pulse Termination EMDW = EM Delayed Wink start EMI = EM Immediate start EMICPD = EM Immediate Start with Calling Party Disconnect EMW = EM Wink start GS = Ground Start GSRB = Ground Start with Reverse Battery LS = Loop Start LSCPD = Loop Start Calling Party Disconnect LSRB = Loop Start with Reverse Battery VoIP = Voice over IP (CMG only) I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. That should be right. If you're really interested in looking, take a cheap voltmeter and put it across the line. If everyone is on hook, you'll see 48V. If someone goes off hook, you'll see it drop to about 6V. If you see a quick drop to 0V when the far end hangs up, you've got battery drop disconnect. I don't know if this has any relevance but I am also originating the clock source from the * side with Wildcard T1 card. That's really the only way it'll work. The channel bank can't generate clocking. ___ 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 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] How to Order Disconnect Supervision from SBC using Adit 600?
I don't know if this has any relevance but I am also originating the clock source from the * side with Wildcard T1 card. That's really the only way it'll work. The channel bank can't generate clocking. Is that a general statement of just confined to the ADIT 600 I thought any T1 interface could either receive timing or it can be a source for timing its just the interface may not be a high quality source eg stratum 4 not stratum1 ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
I don't know if this has any relevance but I am also originating the clock source from the * side with Wildcard T1 card. That's really the only way it'll work. The channel bank can't generate clocking. Are you sure about that? I'm confident that I have both my Adit600 *and* my cheap-ass AB1 providing clocking *to* asterisk. (not both at the same time, but when I was testing with both) Regards, Andrew ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
On Wed, 2004-01-14 at 11:44, TC wrote: I don't know if this has any relevance but I am also originating the clock source from the * side with Wildcard T1 card. That's really the only way it'll work. The channel bank can't generate clocking. Is that a general statement of just confined to the ADIT 600 I thought any T1 interface could either receive timing or it can be a source for timing its just the interface may not be a high quality source eg stratum 4 not stratum1 I'm pretty sure most devices can produce timing, just that channel banks tend to be setup to default to accepting timing. This would after all be the normal config for most sites. The ADIT I'm pretty sure can produce timing, as I think I had mine setup that way once. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
On Wednesday, January 14, 2004 11:45 AM, TC [SMTP:[EMAIL PROTECTED] wrote: I don't know if this has any relevance but I am also originating the clock source from the * side with Wildcard T1 card. That's really the only way it'll work. The channel bank can't generate clocking. Is that a general statement of just confined to the ADIT 600 I thought any T1 interface could either receive timing or it can be a source for timing its just the interface may not be a high quality source eg stratum 4 not stratum1 Any clock can be the source. Of course, you should select the highest available stratum clock and time everything else from this if possible. Another problem is if equipment at both ends of a T1 are trying to derive timing from the other end. This can cause an unstable clock and create synchronization problems. Always, always for T1s, one end needs to provide timing and the other end derive timing. (The only exception is if the equipment on both ends are tied to stratum 1 clocks. However, I would guess that this does not apply to any of us on this list). Don Pobanz ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
I don't know if this has any relevance but I am also originating the clock source from the * side with Wildcard T1 card. That's really the only way it'll work. The channel bank can't generate clocking. be carefull with semantics the transmit leg of the T1 leaving the channel bank is clocked by the channel bank, and therefore could be used by the device at the opposite end for sync. However, the clock within an end-node channel bank is (by design) syncing off the receive leg of the T1. Therefore, by design it is not an official source of clock sync. Words #?!%* Is that a general statement of just confined to the ADIT 600 I thought any T1 interface could either receive timing or it can be a source for timing its just the interface may not be a high quality source eg stratum 4 not stratum1 exactly. ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
I have a little more info on this. Following the suggestion of another post on this topic I tracked down an analog phone with lighted buttons powered by the phone connection. I directly connected the phone to one of my inbound lines and called it with my cell phone. Picked up the analog phone, verified call completion and then hung up my cell. I watched and waited for the lights to go out. Sure enough they did, but it took 8 seconds from the time of the hangup. After the flash more phone started emitting a dialtone sound. Is this correct? I was under the impression the voltage drop would happen almost immediately. Do you have another analog line that's on the same Central office as the line in question? The delay could be lag time betwee the time you hang up your cell phone, the cell provider MTSO processes the hang up, passes it on to their termination provider, who then passes it on to your termination provider, who then passes it on to you. ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
On Tue, 13 Jan 2004, James Sharp wrote: I have a little more info on this. Following the suggestion of another post on this topic I tracked down an analog phone with lighted buttons powered by the phone connection. I directly connected the phone to one of my inbound lines and called it with my cell phone. Picked up the analog phone, verified call completion and then hung up my cell. I watched and waited for the lights to go out. Sure enough they did, but it took 8 seconds from the time of the hangup. After the flash more phone started emitting a dialtone sound. Is this correct? I was under the impression the voltage drop would happen almost immediately. Do you have another analog line that's on the same Central office as the line in question? The delay could be lag time betwee the time you hang up your cell phone, the cell provider MTSO processes the hang up, passes it on to their termination provider, who then passes it on to your termination provider, who then passes it on to you. I will try that, but the results were consistent with what we are seeing across the board. I will call from my pbx to my test line in the morning and time the results. ___ 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
