Re: [asterisk-users] 911 multple-alert question
This is as easy as running an AGI on your 911 rule to do whatever you want. The AGI can dial multiple phones, send emails, page you, etc. Even without the AGI you can do many things from the dialplan. On Sat, 2012-06-09 at 07:51 -0600, Nunya Biznatch wrote: Can you set up asterisk so when a 911 call is placed, in addition to the call out to the PSAP, it also alerts multiple other phones on the switch and will display detailed information. Such as alerting a receptionist or security guard there is a 911 call elsewhere in the building and the location of that call within the building? If so, how? Thanks in advance for the help... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 multple-alert question
On Tue, Jun 12, 2012 at 9:44 AM, Carlos Chavez cur...@telecomabmex.comwrote: This is as easy as running an AGI on your 911 rule to do whatever you want. The AGI can dial multiple phones, send emails, page you, etc. Even without the AGI you can do many things from the dialplan. Here's one example of non-AGI notification: exten = s,n,System(/usr/sbin/sendEmail -t car...@televolve.com -f swit...@televolve.com -u 911 call was placed -m 911 call from ${CDR(accountcode)} ${CALLERID(num)} - ${CALLERID(name)} via switch-1.televolve.com to ${911PROVIDER}.) This of course requires the sendEmail program, which is hugely useful throughout our systems for many purposes. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 multple-alert question
Thanks for the response. You gave me some ideas I didn't think of such as sending a text message to the on-call security person's cell phone. However, while I know I can get the 911 call to call other phones, I also need location data. I know there are ways to do it, but I don't have the kung-fu for things like databases, and am wondering if there's something simple in Asterisk like a flat file used to correlate phone number and location. Then, there's the part of how to get that additional data to display on a phone. To throw a wrench in it, I don't want local security to answer, just to be alerted the call is going on so they can be the first on scene and make themselves available to direct emergency personnel when they arrive. Thanks Again! On 6/12/2012 11:21 AM, Carlos Alvarez wrote: On Tue, Jun 12, 2012 at 9:44 AM, Carlos Chavez cur...@telecomabmex.com mailto:cur...@telecomabmex.com wrote: This is as easy as running an AGI on your 911 rule to do whatever you want. The AGI can dial multiple phones, send emails, page you, etc. Even without the AGI you can do many things from the dialplan. Here's one example of non-AGI notification: exten = s,n,System(/usr/sbin/sendEmail -t car...@televolve.com mailto:car...@televolve.com -f swit...@televolve.com mailto:swit...@televolve.com -u 911 call was placed -m 911 call from ${CDR(accountcode)} ${CALLERID(num)} - ${CALLERID(name)} via switch-1.televolve.com http://switch-1.televolve.com to ${911PROVIDER}.) This of course requires the sendEmail program, which is hugely useful throughout our systems for many purposes. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 multple-alert question
On Tue, 12 Jun 2012, Nunya Biznatch wrote: I also need location data. I know there are ways to do it, but I don't have the kung-fu for things like databases, and am wondering if there's something simple in Asterisk like a flat file used to correlate phone number and location. 1) The Asterisk database? You can access it with dialplan applications. 2) Set the location as a channel variable in sip.conf? (Keeps all of the 'phone specific' stuff in one place. Just being a bit paranoid, but fiddling with 911 calls always makes me nervous. I'd dial the 'real' 911 call over a copper pair first and then after (or in parallel) do all your kewl stuff. I'm sure there will be a lawyer somewhere out there just itching to sue when the 'real' 911 call doesn't happen because of some minor FU on your part. What's your legal exposure if your location data is wrong? I think I'd want a letter absolving me of liability signed by the CEO in my back pocket. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 multple-alert question
You're absolutely correct. The 911 system maintenance is a PITA. Our current PBX routes all 911 calls to an box we call the Proctor Box. That box does a couple things. First, it assigns an ANI to a number. So for example, 3-digit extensions get a real 10-digit numbers assigned. It then routes that 911 call out a dedicated CAMA trunk to the PSAP. Additionally , we pay a third party company to upload our location data to the ALI database, so when our provided ANI hits the PSAP, they can pull proper location info out of the ALI database. The other thing this box does is feed a couple local display terminals. These terminals alarm and display local information from it's own internal database during a 911 call for local security folks. In the meantime, I have the PBX itself which contains its own telephone directory with location and department information, and it's own Emergency Services ID. It will use this ID if the CAMA trunks are out of service and the phone system decides to route the 911 call out a normal PRI. This data we try to use as a baseline for the 911 data. So we currently have multiple databases I have to keep 100% in sync, with a 1000+ set campus with people moving constantly amongst the 25+ buildings. It's a nightmare. Basically, as we migrate to Asterisk, I need to figure out if I can replicate our current functionality. The preference is to come up with something much nicer than a half-dozen data points that are never in synch. If I can't find a solution in Asterisk, then I'm stuck using something like we already have. ...and yes, as you have imagined, there have been a few FUs in the past. We've dodged a number of bullets. I'm hoping I can resolve this when we migrate. I will look into the local channel variable in the sip.conf. That sounds promising. If I populate that data, how does it make it to the display of a phone on campus? I guess that's a piece I haven't either read or been able to wrap my head around yet. Thanks! On 6/12/2012 5:50 PM, Steve Edwards wrote: On Tue, 12 Jun 2012, Nunya Biznatch wrote: I also need location data. I know there are ways to do it, but I don't have the kung-fu for things like databases, and am wondering if there's something simple in Asterisk like a flat file used to correlate phone number and location. 1) The Asterisk database? You can access it with dialplan applications. 2) Set the location as a channel variable in sip.conf? (Keeps all of the 'phone specific' stuff in one place. Just being a bit paranoid, but fiddling with 911 calls always makes me nervous. I'd dial the 'real' 911 call over a copper pair first and then after (or in parallel) do all your kewl stuff. I'm sure there will be a lawyer somewhere out there just itching to sue when the 'real' 911 call doesn't happen because of some minor FU on your part. What's your legal exposure if your location data is wrong? I think I'd want a letter absolving me of liability signed by the CEO in my back pocket. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 multple-alert question
On Tue, 12 Jun 2012, Nunya Biznatch wrote: I will look into the local channel variable in the sip.conf. That sounds promising. If I populate that data, how does it make it to the display of a phone on campus? I guess that's a piece I haven't either read or been able to wrap my head around yet. sip.conf: [my-extension] setvar = LOCATION=Main Lobby extensions.conf exten = *,n,set(CALLERID(name)=Tora! Tora! Tora!) exten = *,n,set(CALLERID(num)=${LOCATION}) exten = *,n,dial(sip/security) exten = *,n,hangup() This shows 'Tora! Tora! Tora!' as line 1 and 'Main Lobby' on my cisco 7960. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 911 multple-alert question
Can you set up asterisk so when a 911 call is placed, in addition to the call out to the PSAP, it also alerts multiple other phones on the switch and will display detailed information. Such as alerting a receptionist or security guard there is a 911 call elsewhere in the building and the location of that call within the building? If so, how? Thanks in advance for the help... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 911, channel full
Hi, I am trying to implement 911 funtionality in my PBX. A call should drop if all lines are busy. Here is my context nineoneone from extensions.conf [nineoneone] exten = s,1,Set(SET_EMERG_FLAG=0) exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten = s,n,Set(EMERGENCY=1,g) exten = s,n,Set(SET_EMERG_FLAG=1) exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress) exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1) exten = s,n,Wait(12) exten = s,n,Goto(checkavail) exten = s,s+2(inprogress),Congestion exten = s,checkavail+101(notavail),Goto(trunkbusy) exten = h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3) exten = h,3,Set(EMERGENCY=0,g) If all lines connecting to PSTN are busy. I get busy tone upon dialing 911 and following message is generated by CLI. app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) I would appreciate if somebody help me solve this issue. Regards Shahnawaz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, channel full
On 3 Mar 2010, at 17:21, mir shahnawaz wrote: [nineoneone] exten = s,1,Set(SET_EMERG_FLAG=0) exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten = s,n,Set(EMERGENCY=1,g) exten = s,n,Set(SET_EMERG_FLAG=1) exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress) exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1) exten = s,n,Wait(12) exten = s,n,Goto(checkavail) exten = s,s+2(inprogress),Congestion exten = s,checkavail+101(notavail),Goto(trunkbusy) exten = h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3) exten = h,3,Set(EMERGENCY=0,g) If all lines connecting to PSTN are busy. I get busy tone upon dialing 911 and following message is generated by CLI. app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) Can you tell us the other lines too? i.e. the bit where it attempts to actually do the hangup.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, channel full
Thanks for your reply. This all I have, am I missing something? Please help in this regard. Here is full output from CLI -- Executing [...@default:1] Goto(SIP/501-0137, nineoneone,s,1) in new stack -- Goto (nineoneone,s,1) -- Executing [...@nineoneone:1] Set(SIP/501-0137, SET_EMERG_FLAG=0) in new stack -- Executing [...@nineoneone:2] ChanIsAvail(SIP/501-0137, DAHDI/g0) in new stack -- Executing [...@nineoneone:3] Set(SIP/501-0137, EMERGENCY=1,g) in new stack -- Executing [...@nineoneone:4] Set(SIP/501-0137, SET_EMERG_FLAG=1) in new stack -- Executing [...@nineoneone:5] Dial(SIP/501-0137, DAHDI/g0/91234567) in new stack [Mar 3 11:26:06] WARNING[28572]: app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/501-0137' status is 'CONGESTION' Regards Shahnawaz On Wed, Mar 3, 2010 at 10:54 AM, Steve Howes steve-li...@geekinter.net wrote: On 3 Mar 2010, at 17:21, mir shahnawaz wrote: [nineoneone] exten = s,1,Set(SET_EMERG_FLAG=0) exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten = s,n,Set(EMERGENCY=1,g) exten = s,n,Set(SET_EMERG_FLAG=1) exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress) exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1) exten = s,n,Wait(12) exten = s,n,Goto(checkavail) exten = s,s+2(inprogress),Congestion exten = s,checkavail+101(notavail),Goto(trunkbusy) exten = h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3) exten = h,3,Set(EMERGENCY=0,g) If all lines connecting to PSTN are busy. I get busy tone upon dialing 911 and following message is generated by CLI. app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) Can you tell us the other lines too? i.e. the bit where it attempts to actually do the hangup.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, Location
Hi Shahnawaz Have you considered how you are going to address location issue for Mobile users calling 911. You should think of SS7 MAP/TCAP to atleast know their Cell ID Regards Sam Thanks very much everybody who contributed their thoughts. I would try to get some DID's so that each physical location can be identified by 911 call Center. Regards Shahnawaz On 2010-01-29, at 2:41 PM, Kevin P. Fleming wrote: Leif Neland wrote: 2: Often callers are answered with an automated message This is 911, please hold, just to give pranksters/misdiallers a chance to hang up before disturbing the operator. Unless 911 records the incoming call right from the start, they will never hear the im-at message. And even if they do, they have to know the message is there to seek on the recording. In the US at least, calls to PSAPs are recorded from the instant the last digit is dialed, before the call is even routed and ringing (on wireline networks where this is possible, anyway). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] 911, location
Thanks very much everybody who contributed their thoughts. I would try to get some DID's so that each physical location can be identified by 911 call Center. Regards Shahnawaz On 2010-01-29, at 2:41 PM, Kevin P. Fleming wrote: Leif Neland wrote: 2: Often callers are answered with an automated message This is 911, please hold, just to give pranksters/misdiallers a chance to hang up before disturbing the operator. Unless 911 records the incoming call right from the start, they will never hear the im-at message. And even if they do, they have to know the message is there to seek on the recording. In the US at least, calls to PSAPs are recorded from the instant the last digit is dialed, before the call is even routed and ringing (on wireline networks where this is possible, anyway). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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] 911, location
Den 28-01-2010 20:15, Danny Nicholas skrev: Here's one solution: - exten = _911,1,Set(IMAT=EXTEN) - exten = _911,2,Set(IMAT=CUT(IMAT|\/|2) - exten = _911,3,Dial(DAHDI/1,w911) - exten = _911,4,Background(emergencyin${IMAT}) Where you would record /var/lib/asterisk/sound/emergencyin100 for extension 100, etc. I see two problems: 1: Doesn't asterisk see a pots-call as answered as soon as it has pressed the last digit and therefore will speak into the ring signal? 2: Often callers are answered with an automated message This is 911, please hold, just to give pranksters/misdiallers a chance to hang up before disturbing the operator. Unless 911 records the incoming call right from the start, they will never hear the im-at message. And even if they do, they have to know the message is there to seek on the recording. An option of the operator receiving a loop of This is a call from the Mickey Mouse building room 123, please press * to receive the call would require the operator to be able to press *, not sure I'd depend my life on that... Leif -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
This might help - exten = _911,1,Set(IMAT=EXTEN) - exten = _911,2,Set(IMAT=CUT(IMAT|\/|2) - exten = _911,3,Dial(DAHDI/1,w911) - exten = _911,4(keepup),Background(emergencyin${IMAT}) - exten = _911,5,wait(10) - exten = _911,6,Goto(keepup) This would repeat the message every 10 seconds... -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Neland Sent: Friday, January 29, 2010 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911, location Den 28-01-2010 20:15, Danny Nicholas skrev: Here's one solution: - exten = _911,1,Set(IMAT=EXTEN) - exten = _911,2,Set(IMAT=CUT(IMAT|\/|2) - exten = _911,3,Dial(DAHDI/1,w911) - exten = _911,4,Background(emergencyin${IMAT}) Where you would record /var/lib/asterisk/sound/emergencyin100 for extension 100, etc. I see two problems: 1: Doesn't asterisk see a pots-call as answered as soon as it has pressed the last digit and therefore will speak into the ring signal? 2: Often callers are answered with an automated message This is 911, please hold, just to give pranksters/misdiallers a chance to hang up before disturbing the operator. Unless 911 records the incoming call right from the start, they will never hear the im-at message. And even if they do, they have to know the message is there to seek on the recording. An option of the operator receiving a loop of This is a call from the Mickey Mouse building room 123, please press * to receive the call would require the operator to be able to press *, not sure I'd depend my life on that... Leif -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
Den 29-01-2010 19:38, Danny Nicholas skrev: This might help - exten = _911,1,Set(IMAT=EXTEN) - exten = _911,2,Set(IMAT=CUT(IMAT|\/|2) - exten = _911,3,Dial(DAHDI/1,w911) - exten = _911,4(keepup),Background(emergencyin${IMAT}) - exten = _911,5,wait(10) - exten = _911,6,Goto(keepup) This would repeat the message every 10 seconds... -- This would prevent the caller talking to the 911-operator, wouldn't it? Leif -- _ -- 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] 911, location
The idea behind the OP was that the caller was a man down who couldn't speak to 911, just dial the number. You could always change wait to waitexten and make an exten to break the loop if you were able to talk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Neland Sent: Friday, January 29, 2010 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911, location Den 29-01-2010 19:38, Danny Nicholas skrev: This might help - exten = _911,1,Set(IMAT=EXTEN) - exten = _911,2,Set(IMAT=CUT(IMAT|\/|2) - exten = _911,3,Dial(DAHDI/1,w911) - exten = _911,4(keepup),Background(emergencyin${IMAT}) - exten = _911,5,wait(10) - exten = _911,6,Goto(keepup) This would repeat the message every 10 seconds... -- This would prevent the caller talking to the 911-operator, wouldn't it? Leif -- _ -- 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 -- _ -- 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] 911, location
Leif Neland wrote: 2: Often callers are answered with an automated message This is 911, please hold, just to give pranksters/misdiallers a chance to hang up before disturbing the operator. Unless 911 records the incoming call right from the start, they will never hear the im-at message. And even if they do, they have to know the message is there to seek on the recording. In the US at least, calls to PSAPs are recorded from the instant the last digit is dialed, before the call is even routed and ringing (on wireline networks where this is possible, anyway). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] 911, location
Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I think its possible by having DID's but I am looking for other options too. I would appreciate your valuable ideas and suggestions. Thanks in advance Shahnawaz -- _ -- 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] 911, location
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mir shahnawaz wrote: Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I think its possible by having DID's but I am looking for other options too. I would appreciate your valuable ideas and suggestions. If you're using POTS lines to make the call to 911 they'll know the location, if the POTS lines come into the building that you're calling from. Are you saying that these lines are located in a different location? Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR UnWTQQ1anTXtDqfk54QVj/k= =LtAE -END PGP SIGNATURE- -- _ -- 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] 911, location
Thanks for your reply. Yes POTS lines are coming into the building but I have multiple rooms. Suppose a person is working late hours and have a heart attack. How could 911 locate the room when no or few people around. Thanks Smir On Thu, Jan 28, 2010 at 11:46 AM, Barry L. Kline blkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mir shahnawaz wrote: Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I think its possible by having DID's but I am looking for other options too. I would appreciate your valuable ideas and suggestions. If you're using POTS lines to make the call to 911 they'll know the location, if the POTS lines come into the building that you're calling from. Are you saying that these lines are located in a different location? Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR UnWTQQ1anTXtDqfk54QVj/k= =LtAE -END PGP SIGNATURE- -- _ -- 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 -- _ -- 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] 911, location
Here's one solution: - exten = _911,1,Set(IMAT=EXTEN) - exten = _911,2,Set(IMAT=CUT(IMAT|\/|2) - exten = _911,3,Dial(DAHDI/1,w911) - exten = _911,4,Background(emergencyin${IMAT}) Where you would record /var/lib/asterisk/sound/emergencyin100 for extension 100, etc. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mir shahnawaz Sent: Thursday, January 28, 2010 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911, location Thanks for your reply. Yes POTS lines are coming into the building but I have multiple rooms. Suppose a person is working late hours and have a heart attack. How could 911 locate the room when no or few people around. Thanks Smir On Thu, Jan 28, 2010 at 11:46 AM, Barry L. Kline blkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mir shahnawaz wrote: Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I think its possible by having DID's but I am looking for other options too. I would appreciate your valuable ideas and suggestions. If you're using POTS lines to make the call to 911 they'll know the location, if the POTS lines come into the building that you're calling from. Are you saying that these lines are located in a different location? Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR UnWTQQ1anTXtDqfk54QVj/k= =LtAE -END PGP SIGNATURE- -- _ -- 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 -- _ -- 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 -- _ -- 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] 911, location
You should phone up the emergency people on a non-emergency number and ask them about that as well. On Thu, Jan 28, 2010 at 10:58 AM, mir shahnawaz shahnawaz...@gmail.com wrote: Thanks for your reply. Yes POTS lines are coming into the building but I have multiple rooms. Suppose a person is working late hours and have a heart attack. How could 911 locate the room when no or few people around. Thanks Smir On Thu, Jan 28, 2010 at 11:46 AM, Barry L. Kline blkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mir shahnawaz wrote: Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I think its possible by having DID's but I am looking for other options too. I would appreciate your valuable ideas and suggestions. If you're using POTS lines to make the call to 911 they'll know the location, if the POTS lines come into the building that you're calling from. Are you saying that these lines are located in a different location? Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR UnWTQQ1anTXtDqfk54QVj/k= =LtAE -END PGP SIGNATURE- -- _ -- 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 -- _ -- 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 -- _ -- 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] 911, location
mir shahnawaz wrote: Thanks, Could you please explain this little bit more. I am not getting IMAT=EXTEN. On Thu, Jan 28, 2010 at 12:15 PM, Danny Nicholasda...@debsinc.com wrote: Here's one solution: - exten = _911,1,Set(IMAT=EXTEN) He probably meant ${EXTEN} Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
Thanks, Could you please explain this little bit more. I am not getting IMAT=EXTEN. On Thu, Jan 28, 2010 at 12:15 PM, Danny Nicholas da...@debsinc.com wrote: Here's one solution: - exten = _911,1,Set(IMAT=EXTEN) - exten = _911,2,Set(IMAT=CUT(IMAT|\/|2) - exten = _911,3,Dial(DAHDI/1,w911) - exten = _911,4,Background(emergencyin${IMAT}) Where you would record /var/lib/asterisk/sound/emergencyin100 for extension 100, etc. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mir shahnawaz Sent: Thursday, January 28, 2010 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911, location Thanks for your reply. Yes POTS lines are coming into the building but I have multiple rooms. Suppose a person is working late hours and have a heart attack. How could 911 locate the room when no or few people around. Thanks Smir On Thu, Jan 28, 2010 at 11:46 AM, Barry L. Kline blkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mir shahnawaz wrote: Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I think its possible by having DID's but I am looking for other options too. I would appreciate your valuable ideas and suggestions. If you're using POTS lines to make the call to 911 they'll know the location, if the POTS lines come into the building that you're calling from. Are you saying that these lines are located in a different location? Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR UnWTQQ1anTXtDqfk54QVj/k= =LtAE -END PGP SIGNATURE- -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] 911, location
- Doug Lytle supp...@drdos.info wrote: mir shahnawaz wrote: Thanks, Could you please explain this little bit more. I am not getting IMAT=EXTEN. On Thu, Jan 28, 2010 at 12:15 PM, Danny Nicholasda...@debsinc.com wrote: Here's one solution: - exten = _911,1,Set(IMAT=EXTEN) He probably meant ${EXTEN} Doug If nobody is around how would they even get into the building ? Certainly in the UK nobody should ever be in the building on their own for this exact reason; and if they are then in would be prudent to have man down alarms and paging. -- Thanks, Phil -- _ -- 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] 911 via MAX TNT ??
Any suggestions ? Available options for the two settings similiar to the one identified are as follows: admin set send-dnis-type-of-number? send-dnis-type-of-number: Type of Number to be sent in called party IE in the setup message to pstn. For ISDN signaling. To be used on egress gateway for VoIP calls. Enumerated field, values: unknown: international: national: network-spec: subscriber: abbreviated: transparent: Setting this, we can pass TON transparently as received from upper layers or in case of VoIP, as received from Near End gateway. admin set send-dnis-numbering-plan? send-dnis-numbering-plan: Numbering Plan to be sent in called party IE in the setup message to pstn. For ISDN signaling. To be used on egress gateway for VoIP calls. Enumerated field, values: unknown: isdn-telephony: data: telex: national: private: transparent: Setting this, we can pass NP transparently as received from upper layers or in case of VoIP, as received from Near End gateway. From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Leon Sun [EMAIL PROTECTED] Sent: Monday, June 09, 2008 1:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 911 via MAX TNT ?? It should work. Leon Sun Times Telecom Tel: 604-279-8787 ext 1586 Fax: 604-278-2793 Mobile: 604-780-2668 Click this button now and leave your phone number. Talk to me for free. powered by www.clicksaya.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Sunday, June 08, 2008 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? We are providing voip services, these 911 calls are going out from our subscribers to the lec to be delivered to the 911 PSAP.. Would this apply in that scenario ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leon Sun Sent: Sunday, June 08, 2008 3:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 911 via MAX TNT ?? Joe, I am not sure if your 911 call is incoming or outgoing on PRIs. #assume you have a T1 in {1 1 1} Read t1 { 1 1 1} Set line send-dnis-type-of-number ? You will see options. Some 911 providers support media-before-connect. Plz make sure your all of TNT support 183. Hope it can help you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Sunday, June 08, 2008 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Alex.. would you point us in the right direction, or perhaps consider sending a sample max tnt config reflecting how this is done? Thank you.. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, June 06, 2008 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? I believe the ISDN call plan can be configured as part of the trunk group / route. Joe Carroll wrote: We talked with the LEC and discovered that 911 has to be sent as Unknown instead of National... Anyone know how we might tell the TNT to do this? Apparently, according to the carrier, all Special Access Numbers, 411, 611, 911, etc require this special code ??? PRI DEBUG FOLLOWS: --nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 Bearer_Cap 80 90 A2 (Speech,Rate=64K) Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) Calling_Num (National,Restricted,Failed) 229317 Called_Num (National) 911 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Thursday, June 05, 2008 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk
Re: [asterisk-users] 911 via MAX TNT ??
It should work. Leon Sun Times Telecom Tel: 604-279-8787 ext 1586 Fax: 604-278-2793 Mobile: 604-780-2668 Click this button now and leave your phone number. Talk to me for free. powered by www.clicksaya.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Sunday, June 08, 2008 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? We are providing voip services, these 911 calls are going out from our subscribers to the lec to be delivered to the 911 PSAP.. Would this apply in that scenario ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leon Sun Sent: Sunday, June 08, 2008 3:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 911 via MAX TNT ?? Joe, I am not sure if your 911 call is incoming or outgoing on PRIs. #assume you have a T1 in {1 1 1} Read t1 { 1 1 1} Set line send-dnis-type-of-number ? You will see options. Some 911 providers support media-before-connect. Plz make sure your all of TNT support 183. Hope it can help you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Sunday, June 08, 2008 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Alex.. would you point us in the right direction, or perhaps consider sending a sample max tnt config reflecting how this is done? Thank you.. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, June 06, 2008 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? I believe the ISDN call plan can be configured as part of the trunk group / route. Joe Carroll wrote: We talked with the LEC and discovered that 911 has to be sent as Unknown instead of National... Anyone know how we might tell the TNT to do this? Apparently, according to the carrier, all Special Access Numbers, 411, 611, 911, etc require this special code ??? PRI DEBUG FOLLOWS: --nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 Bearer_Cap 80 90 A2 (Speech,Rate=64K) Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) Calling_Num (National,Restricted,Failed) 229317 Called_Num (National) 911 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Thursday, June 05, 2008 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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 ___ -- 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 ___ -- 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 -- Alex Balashov Evariste Systems Web
Re: [asterisk-users] 911 via MAX TNT ??
Alex.. would you point us in the right direction, or perhaps consider sending a sample max tnt config reflecting how this is done? Thank you.. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, June 06, 2008 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? I believe the ISDN call plan can be configured as part of the trunk group / route. Joe Carroll wrote: We talked with the LEC and discovered that 911 has to be sent as Unknown instead of National... Anyone know how we might tell the TNT to do this? Apparently, according to the carrier, all Special Access Numbers, 411, 611, 911, etc require this special code ??? PRI DEBUG FOLLOWS: --nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 Bearer_Cap 80 90 A2 (Speech,Rate=64K) Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) Calling_Num (National,Restricted,Failed) 229317 Called_Num (National) 911 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Thursday, June 05, 2008 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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 ___ -- 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 ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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 ___ -- 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] 911 via MAX TNT ??
Joe, I am not sure if your 911 call is incoming or outgoing on PRIs. #assume you have a T1 in {1 1 1} Read t1 { 1 1 1} Set line send-dnis-type-of-number ? You will see options. Some 911 providers support media-before-connect. Plz make sure your all of TNT support 183. Hope it can help you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Sunday, June 08, 2008 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Alex.. would you point us in the right direction, or perhaps consider sending a sample max tnt config reflecting how this is done? Thank you.. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, June 06, 2008 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? I believe the ISDN call plan can be configured as part of the trunk group / route. Joe Carroll wrote: We talked with the LEC and discovered that 911 has to be sent as Unknown instead of National... Anyone know how we might tell the TNT to do this? Apparently, according to the carrier, all Special Access Numbers, 411, 611, 911, etc require this special code ??? PRI DEBUG FOLLOWS: --nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 Bearer_Cap 80 90 A2 (Speech,Rate=64K) Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) Calling_Num (National,Restricted,Failed) 229317 Called_Num (National) 911 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Thursday, June 05, 2008 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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 ___ -- 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 ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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 ___ -- 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 ___ -- 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] 911 via MAX TNT ??
We are providing voip services, these 911 calls are going out from our subscribers to the lec to be delivered to the 911 PSAP.. Would this apply in that scenario ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leon Sun Sent: Sunday, June 08, 2008 3:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 911 via MAX TNT ?? Joe, I am not sure if your 911 call is incoming or outgoing on PRIs. #assume you have a T1 in {1 1 1} Read t1 { 1 1 1} Set line send-dnis-type-of-number ? You will see options. Some 911 providers support media-before-connect. Plz make sure your all of TNT support 183. Hope it can help you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Sunday, June 08, 2008 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Alex.. would you point us in the right direction, or perhaps consider sending a sample max tnt config reflecting how this is done? Thank you.. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, June 06, 2008 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? I believe the ISDN call plan can be configured as part of the trunk group / route. Joe Carroll wrote: We talked with the LEC and discovered that 911 has to be sent as Unknown instead of National... Anyone know how we might tell the TNT to do this? Apparently, according to the carrier, all Special Access Numbers, 411, 611, 911, etc require this special code ??? PRI DEBUG FOLLOWS: --nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 Bearer_Cap 80 90 A2 (Speech,Rate=64K) Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) Calling_Num (National,Restricted,Failed) 229317 Called_Num (National) 911 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Thursday, June 05, 2008 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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 ___ -- 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 ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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 ___ -- Bandwidth and Colocation Provided by http
Re: [asterisk-users] 911 via MAX TNT ??
We talked with the LEC and discovered that 911 has to be sent as Unknown instead of National... Anyone know how we might tell the TNT to do this? Apparently, according to the carrier, all Special Access Numbers, 411, 611, 911, etc require this special code ??? PRI DEBUG FOLLOWS: --nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 Bearer_Cap 80 90 A2 (Speech,Rate=64K) Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) Calling_Num (National,Restricted,Failed) 229317 Called_Num (National) 911 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Thursday, June 05, 2008 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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 ___ -- 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 ___ -- 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] 911 via MAX TNT ??
I believe the ISDN call plan can be configured as part of the trunk group / route. Joe Carroll wrote: We talked with the LEC and discovered that 911 has to be sent as Unknown instead of National... Anyone know how we might tell the TNT to do this? Apparently, according to the carrier, all Special Access Numbers, 411, 611, 911, etc require this special code ??? PRI DEBUG FOLLOWS: --nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 Bearer_Cap 80 90 A2 (Speech,Rate=64K) Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) Calling_Num (National,Restricted,Failed) 229317 Called_Num (National) 911 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Thursday, June 05, 2008 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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 ___ -- 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 ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] 911 via MAX TNT ??
On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] 911 via MAX TNT ??
Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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 ___ -- 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] 911 via MAX TNT
When I send a call out the MAX I get the following -- Got SIP response 484 Address Incomplete back from 172.16.10.230 Any ideas on how to make 911 appear as a ten digit number to the device so that it will pass the number out to the PSTN ? This is not a max tnt problem, the tnt will pass anything you send to it, 911/411/7 digit/10digit/011 international, the question is, does your PSTN provider accept 911 call on the trunk your passing the call to? JR ___ -- 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] 911 via MAX TNT ??
On 6/3/08, Joe Carroll [EMAIL PROTECTED] wrote: Quick question for the folks using MAX TNTs for aggregators.. When I send a call out the MAX I get the following…. -- Got SIP response 484 Address Incomplete back from 172.16.10.230 Any ideas on how to make 911 appear as a ten digit number to the device so that it will pass the number out to the PSTN ? I've never used a TNT before but what does your dial pattern matching look like? If you were using Asterisk, your match would probably look like this: _NXXNXX 911 would match NXX but not the remaining digits, hence the 484 Address Incomplete. I bet your TNT is doing something similar However: _NXXNXX _911 Would work just fine (in Asterisk). You need to figure out how to do something similar on your TNT. -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- 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] 911 via MAX TNT ??
The first place you may want to look is in the SYSLOG of the TNT, allowing you to see things such as the ISDN error code along with the SIP code. You can try to catch that on the terminal of the TNT, but it may make more sense to pipe your syslogs out to an external box, if you aren't doing it already. JR's suggestion that it may be a limit of the trunk you're using. Joe Carroll wrote: Hi Mik: The TNT is at the ip address 172.16.10.230 and the asterisk box is at 172.16.10.240... The trunk group is 3100.. so we send 3100911 to the TNT and get the message below.. I couldn't figure it out..any ideas ?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mik Cheez Sent: Tuesday, June 03, 2008 8:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Without knowing more about how you have your TNT set up, typically you'd configure your outbound T1's to a specific trunkgroup and prepend that trunkgroup number to the phonenumber. Should it be assumed that 172.16.10.230 is the address of the TNT? Mik Joe Carroll wrote: Quick question for the folks using MAX TNTs for aggregators.. When I send a call out the MAX I get the following -- Got SIP response 484 Address Incomplete back from 172.16.10.230 Any ideas on how to make 911 appear as a ten digit number to the device so that it will pass the number out to the PSTN ? ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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] 911 via MAX TNT ??
See below, we replaced the area code and prefix of with NPANXX for concerns Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 Extension Changed NPANXX7604 new state InUse for Notify User NPANXX7555 -- Executing [EMAIL PROTECTED]:1] Set(SIP/NPANXX7604-08c46518, CALLERID(number)=NPANXX3551) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/NPANXX7604-08c46518, SIP/To-TNT/3100911) in new stack -- Called To-TNT/3100911 Really destroying SIP dialog '[EMAIL PROTECTED]' Method: NOTIFY -- Got SIP response 484 Address Incomplete back from 172.16.10.230 == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/NPANXX7604-08c46518' status is 'CHANUNAVAIL' -- Executing [EMAIL PROTECTED]:1] Set(SIP/NPANXX7604-08c46518, CDR(userfield)=) in new stack Extension Changed NXX5557604 new state Idle for Notify User NXX5557555 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mik Cheez Sent: Wednesday, June 04, 2008 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? The first place you may want to look is in the SYSLOG of the TNT, allowing you to see things such as the ISDN error code along with the SIP code. You can try to catch that on the terminal of the TNT, but it may make more sense to pipe your syslogs out to an external box, if you aren't doing it already. JR's suggestion that it may be a limit of the trunk you're using. ___ -- 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] 911 via MAX TNT ??
Cause 28 indicates Invalid number format. Joe Carroll wrote: See below, we replaced the area code and prefix of with NPANXX for concerns Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 Extension Changed NPANXX7604 new state InUse for Notify User NPANXX7555 -- Executing [EMAIL PROTECTED]:1] Set(SIP/NPANXX7604-08c46518, CALLERID(number)=NPANXX3551) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/NPANXX7604-08c46518, SIP/To-TNT/3100911) in new stack -- Called To-TNT/3100911 Really destroying SIP dialog '[EMAIL PROTECTED]' Method: NOTIFY -- Got SIP response 484 Address Incomplete back from 172.16.10.230 == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/NPANXX7604-08c46518' status is 'CHANUNAVAIL' -- Executing [EMAIL PROTECTED]:1] Set(SIP/NPANXX7604-08c46518, CDR(userfield)=) in new stack Extension Changed NXX5557604 new state Idle for Notify User NXX5557555 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mik Cheez Sent: Wednesday, June 04, 2008 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? The first place you may want to look is in the SYSLOG of the TNT, allowing you to see things such as the ISDN error code along with the SIP code. You can try to catch that on the terminal of the TNT, but it may make more sense to pipe your syslogs out to an external box, if you aren't doing it already. JR's suggestion that it may be a limit of the trunk you're using. ___ -- 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 ___ -- 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] 911 via MAX TNT ??
On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... -A. ___ -- 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] 911 versus 9.911
I know every second counts in a real 911 situation, but what about adding a pause in the call flow. Maybe a 1 second pause before actually passing the digits to the provider. This gives the user 1 second to realize the mistake and hang up, longer than 1 seconds is a real emergency. Just a thought. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Aarons (US) Sent: Wednesday, August 30, 2006 10:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 911 versus 9.911 Is there a FCC or other North America requirement that I provide 911 versus 9.911. I want to require users to dial 9.911 in our office, and remove 911. Are there any statutory requirements or laws about this? User accidentially dial 9 then 1 then another 1 and hangup. Weve educated them to stay on the line and ever hang up, but they hang up anyway, resulting in fines for excess hangups to 911. Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 versus 9.911
I have enabled outside extension '911' and '11' for emergency service. This way users can either dial '9911' or '911' to get to a PSAP. I would rather have a couple accidential 911 calls than a death because someone forgot to dial a 2nd 9. When people are freaking out they fall back on muscle memory. Many won't pay attention to wether they need to dial 9 for an outside line and then 911 for emergency. I know every second counts in a real 911 situation, but what about adding a pause in the call flow. Maybe a 1 second pause before actually passing the digits to the provider. This gives the user 1 second to realize the mistake and hang up, longer than 1 seconds is a real emergency. Just a thought. From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Jason Aarons (US) Sent: Wednesday, August 30, 2006 10:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 911 versus 9.911 Is there a FCC or other North America requirement that I provide 911 versus 9.911. I want to require users to dial 9.911 in our office, and remove 911. Are there any statutory requirements or laws about this? User accidentially dial 9 then 1 then another 1 and hangup. We’ve educated them to stay on the line and ever hang up, but they hang up anyway, resulting in fines for excess hangups to 911. Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 versus 9.911
I play a recording that starts as soon as the second 1 is pressed: If this is an emergency, please hang up and dial 9-911. Short, simple, and to the point. -- Strom Carlson http://www.stromcarlson.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 versus 9.911
I once worked for a big accounting firm who eliminated this problem very simply -- they used 7 to get a trunk. 7911 and 911 would still get you an emergency operator, but accidental 911 calls were all but a thing of the past. Aaron Daniel wrote: On Wed, 2006-08-30 at 20:10 -0700, George Pajari wrote: I'd rather pay the fine than the liability settlement when found negligent in a lawsuit because someone panicked, repeatedly dialled 911, and could not reach Emergency when their coworker had a major myocardial infarction right beside them. We configure all our systems, regardless of whether or not they have a dial-9 for an outside line dialplan, to route both 911 and 9911 to an outside line and 911. We also log every call so when someone does dial and hangup, we send Big Eric to their cube to rearrange a few fingers on their dialling hand :-) Most people are going to attempt to dial 911 regardless of where they are, especially if they're in a panic... We use both 911 and 9911 (our nortel expects 9911, but allows 911), which seems to be better for users that aren't used to the dial 9 to get out mentality. 100 accidental calls is worth the 1 time that someone could die because they don't realize that they're supposed to dial 9911 instead. We've actually had on several occasions people in my office dial 911 on accident when dialing something like 91800. and ended up hitting the 1 twice, and usually dispatch just calls back and asks what was up. Just my 2 cents. Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 911 versus 9.911
Is there a FCC or other North America requirement that I provide 911 versus 9.911. I want to require users to dial 9.911 in our office, and remove 911. Are there any statutory requirements or laws about this? User accidentially dial 9 then 1 then another 1 and hangup. Weve educated them to stay on the line and ever hang up, but they hang up anyway, resulting in fines for excess hangups to 911. Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 versus 9.911
Jason Aarons (US) wrote: Is there a FCC or other North America requirement that I provide 911 versus 9.911. I want to require users to dial 9.911 in our office, and remove 911. Are there any statutory requirements or laws about this? User accidentially dial 9 then 1 then another 1 and hangup. We’ve educated them to stay on the line and ever hang up, but they hang up anyway, resulting in fines for excess hangups to 911. I'd rather pay the fine than the liability settlement when found negligent in a lawsuit because someone panicked, repeatedly dialled 911, and could not reach Emergency when their coworker had a major myocardial infarction right beside them. We configure all our systems, regardless of whether or not they have a dial-9 for an outside line dialplan, to route both 911 and 9911 to an outside line and 911. We also log every call so when someone does dial and hangup, we send Big Eric to their cube to rearrange a few fingers on their dialling hand :-) g. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 versus 9.911
On Wed, 2006-08-30 at 20:10 -0700, George Pajari wrote: I'd rather pay the fine than the liability settlement when found negligent in a lawsuit because someone panicked, repeatedly dialled 911, and could not reach Emergency when their coworker had a major myocardial infarction right beside them. We configure all our systems, regardless of whether or not they have a dial-9 for an outside line dialplan, to route both 911 and 9911 to an outside line and 911. We also log every call so when someone does dial and hangup, we send Big Eric to their cube to rearrange a few fingers on their dialling hand :-) Most people are going to attempt to dial 911 regardless of where they are, especially if they're in a panic... We use both 911 and 9911 (our nortel expects 9911, but allows 911), which seems to be better for users that aren't used to the dial 9 to get out mentality. 100 accidental calls is worth the 1 time that someone could die because they don't realize that they're supposed to dial 9911 instead. We've actually had on several occasions people in my office dial 911 on accident when dialing something like 91800. and ended up hitting the 1 twice, and usually dispatch just calls back and asks what was up. Just my 2 cents. Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 Testing
In my case the 911 goes to the Police departments dispatchers, I have to to call the main office number and make sure it is a good time to test then I can call right back to get the read out on the screen. This is great since they want me to send them some sort of information about who called from my system. On 8/13/06, John Novack [EMAIL PROTECTED] wrote: Petty bureaucrats make up their own rules the world over!Whatever you do, it will be wrongJohn NovackKevin Kiely wrote: Be careful here... Our local PSAP is handled by the fire department.I had one of our guy's make a test call and we were told that this test must be coordinated and scheduled in advance with the chief.They want no test calls.It would probable be safest to check before making the call as they could consider it an abuse of the emergency system.It seems like a catch 22. -Original Message- From: Shane Young [mailto: [EMAIL PROTECTED]] Sent: Sunday, August 13, 2006 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Leif Neland Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 Testing Quoting Leif Neland [EMAIL PROTECTED]: According to what I've read somewhere, at least our 911 (112) has an answering machine, saying Alarm central, one moment and a few seconds delay, before the call actually is signaled to the dispatcher, to filter out misdials and crank calls. So if you hang up quickly, they'll never know or be bothered. In Minnesota (probably most places in the US) Once you have dialed 911, even if it was in error, you should stay on the line until a dispatcher answers.If you don't they'll consider it a 911 hangup and attempt to call you back.If they can not reach you, they will dispatch a law enforcement officer (and in some areas, other emergency services). The usual call flow I've experianced is this: I Dial 911 They answer Minneapolis 911 I say This is Shane from company x making a 911 test call. They will either say ok or Please Hold if they have other calls waiting. Once they have said ok, I'll say I want to confirm you see my number as xxx-xxx- and my address is y They will almost always say Yes, that's what we have I'll say Thank you They will say Good Bye and hang up. I'll hang up. --Shane This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 911 Testing
Good MorningList, When setting up a pbx and you want to test your 911 settings do you call 911 and tell them its a test call or do you relly that you set it up properly and hope for the best when some one call's 911 ? Thanks, Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 Testing
Dovid Bender wrote: Good Morning List, When setting up a pbx and you want to test your 911 settings do you call 911 and tell them its a test call or do you relly that you set it up properly and hope for the best when some one call's 911 ? I believe most 911 centers would prefer you call their non-emergency number before testing to let them know what you're about to do. They may suggest a less busy time to do the tests, etc. I know a lot of installers that just dial 911 without any previous contact and I don't recall any of them getting chewed out for doing it. Guess if you keep the conversation short its less likely to be a bother. From a quality control perspective, I'm sure they would appreciate a test call to ensure caller id / ani, street address, etc, are accurate. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 Testing
Rich Adamson wrote: Dovid Bender wrote: Good Morning List, When setting up a pbx and you want to test your 911 settings do you call 911 and tell them its a test call or do you relly that you set it up properly and hope for the best when some one call's 911 ? I believe most 911 centers would prefer you call their non-emergency number before testing to let them know what you're about to do. They may suggest a less busy time to do the tests, etc. I know a lot of installers that just dial 911 without any previous contact and I don't recall any of them getting chewed out for doing it. Guess if you keep the conversation short its less likely to be a bother. According to what I've read somewhere, at least our 911 (112) has an answering machine, saying Alarm central, one moment and a few seconds delay, before the call actually is signaled to the dispatcher, to filter out misdials and crank calls. So if you hang up quickly, they'll never know or be bothered. Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 Testing
Quoting Leif Neland [EMAIL PROTECTED]: According to what I've read somewhere, at least our 911 (112) has an answering machine, saying Alarm central, one moment and a few seconds delay, before the call actually is signaled to the dispatcher, to filter out misdials and crank calls. So if you hang up quickly, they'll never know or be bothered. In Minnesota (probably most places in the US) Once you have dialed 911, even if it was in error, you should stay on the line until a dispatcher answers. If you don't they'll consider it a 911 hangup and attempt to call you back. If they can not reach you, they will dispatch a law enforcement officer (and in some areas, other emergency services). The usual call flow I've experianced is this: I Dial 911 They answer Minneapolis 911 I say This is Shane from company x making a 911 test call. They will either say ok or Please Hold if they have other calls waiting. Once they have said ok, I'll say I want to confirm you see my number as xxx-xxx- and my address is y They will almost always say Yes, that's what we have I'll say Thank you They will say Good Bye and hang up. I'll hang up. --Shane This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 911 Testing
Be careful here... Our local PSAP is handled by the fire department. I had one of our guy's make a test call and we were told that this test must be coordinated and scheduled in advance with the chief. They want no test calls. It would probable be safest to check before making the call as they could consider it an abuse of the emergency system. It seems like a catch 22. -Original Message- From: Shane Young [mailto:[EMAIL PROTECTED] Sent: Sunday, August 13, 2006 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Leif Neland Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 Testing Quoting Leif Neland [EMAIL PROTECTED]: According to what I've read somewhere, at least our 911 (112) has an answering machine, saying Alarm central, one moment and a few seconds delay, before the call actually is signaled to the dispatcher, to filter out misdials and crank calls. So if you hang up quickly, they'll never know or be bothered. In Minnesota (probably most places in the US) Once you have dialed 911, even if it was in error, you should stay on the line until a dispatcher answers. If you don't they'll consider it a 911 hangup and attempt to call you back. If they can not reach you, they will dispatch a law enforcement officer (and in some areas, other emergency services). The usual call flow I've experianced is this: I Dial 911 They answer Minneapolis 911 I say This is Shane from company x making a 911 test call. They will either say ok or Please Hold if they have other calls waiting. Once they have said ok, I'll say I want to confirm you see my number as xxx-xxx- and my address is y They will almost always say Yes, that's what we have I'll say Thank you They will say Good Bye and hang up. I'll hang up. --Shane This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.10.9/417 - Release Date: 8/11/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 Testing
Petty bureaucrats make up their own rules the world over! Whatever you do, it will be wrong John Novack Kevin Kiely wrote: Be careful here... Our local PSAP is handled by the fire department. I had one of our guy's make a test call and we were told that this test must be coordinated and scheduled in advance with the chief. They want no test calls. It would probable be safest to check before making the call as they could consider it an abuse of the emergency system. It seems like a catch 22. -Original Message- From: Shane Young [mailto:[EMAIL PROTECTED] Sent: Sunday, August 13, 2006 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Leif Neland Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 Testing Quoting Leif Neland [EMAIL PROTECTED]: According to what I've read somewhere, at least our 911 (112) has an answering machine, saying Alarm central, one moment and a few seconds delay, before the call actually is signaled to the dispatcher, to filter out misdials and crank calls. So if you hang up quickly, they'll never know or be bothered. In Minnesota (probably most places in the US) Once you have dialed 911, even if it was in error, you should stay on the line until a dispatcher answers. If you don't they'll consider it a 911 hangup and attempt to call you back. If they can not reach you, they will dispatch a law enforcement officer (and in some areas, other emergency services). The usual call flow I've experianced is this: I Dial 911 They answer Minneapolis 911 I say This is Shane from company x making a 911 test call. They will either say ok or Please Hold if they have other calls waiting. Once they have said ok, I'll say I want to confirm you see my number as xxx-xxx- and my address is y They will almost always say Yes, that's what we have I'll say Thank you They will say Good Bye and hang up. I'll hang up. --Shane This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 @ Zap Channel Breakin
On Sun, May 14, 2006 at 01:01:49PM -1000, Mark Coccimiglio wrote: [...] disconnected. Then immediately drop the call capture the line so noone else can use it, wait about 5 seconds for the telco to clear the far end and place the 911 call. Is this possible? How about this: http://www.voip-info.org/wiki/view/Asterisk+tips+911 -- -Will :: AD6XL Orton :: http://www.loopfree.net/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 911 @ Zap Channel Breakin
Ok here is one for you. I know we all do the this for 911: exten = _911,1,Dial(Zap/1/911) exten = _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a least cost routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is durring a 911 call test if the Zap channel is Available (probably using ChanIsAvail() ) to test the line. IF the channel is in use I want to barge in with an announcment saying that the line is needed for an emergency and the call we be disconnected. Then immediately drop the call capture the line so noone else can use it, wait about 5 seconds for the telco to clear the far end and place the 911 call. Is this possible? Thnaks Mark C [EMAIL PROTECTED] FWD: 293625 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 @ Zap Channel Breakin
Mark Coccimiglio wrote: Ok here is one for you. I know we all do the this for 911: exten = _911,1,Dial(Zap/1/911) exten = _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a least cost routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is durring a 911 call test if the Zap channel is Available (probably using ChanIsAvail() ) to test the line. IF the channel is in use I want to barge in with an announcment saying that the line is needed for an emergency and the call we be disconnected. Then immediately drop the call capture the line so noone else can use it, wait about 5 seconds for the telco to clear the far end and place the 911 call. Is this possible? Thnaks Mark C [EMAIL PROTECTED] FWD: 293625 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I seem to recall a script that did something similar on voip-info.org. Look through their E911 stuff, perhaps it's still there. Good luck. -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 911 and ISDN PRI
Title: Message It looks like the outbound caller ID is not being set properly. Most of the carriers that I've dealt with will act exactly as you said if you do not set it to what is expected at the 911 center. In particular: Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3251' ] Your user number being sent is just the caller ID of the SIP channel. Regards, - Brad -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe PukepailSent: Tuesday, February 07, 2006 3:26 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 911 and ISDN PRI I have a call in with the carrier, below is the PRI debug, looks like it is getting hungup because of "Invalid Number format", I did try to use Setcallerid to change the callerID to a DID number in a previous attempt, but it still didn't go through. Not sure if that "invalid number format" is the calling number or the number I'm calling. I'll let the list know the result. I would encourage everyone to test their 911 functionality (especially if you have a PRI), I almost didn't check it.The PRI is up and 411 works, so I almost assumed that 911 would work. Make sure you call the police station first to make sure they are not swamped by real emergency calls and let them know you are testing . -- Executing NoOp("SIP/3251-7316", "3251") in new stack -- Executing Dial("SIP/3251-7316", "Zap/g2/911") in new stack-- Making new call for cr 33144 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=44 Call Ref: len= 2 (reference 376/0x178) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 09 b1 52 65 63 70 74 69 6f 6e] Display (len= 9) Charset: 31 [ Recption ] [6c 06 41 80 33 32 35 31] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3251' ] [70 04 a1 39 31 31] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '911' ] -- Called g2/911 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 376/0x178) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 9c] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ]-- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangup On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote: I dunno about your provider but I know that 2 of my 3 MCI PRI circuitshave no 911 abilities. MCI tells me this is becasue I have no local dialing plan on them.Mark, G7LTT/KC2ENIRandolph, NJhttp://www.g7ltt.comMichael Collins wrote: 911 **should** work on a PRI.If you are getting a hangup and you don't see a valid hangupcause, it might be best to get your carrier on the line and have them monitor the circuit while you dial 911.They might be able to tell you what the problem is. -MC *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] *On Behalf Of *Joe Pukepail *Sent:* Tuesday, February 07, 2006 10:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] 911 and ISDN PRI Does asterisk support this?I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup.Does 911 normally work over a PRI line?Anything special I have to setup in asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___
Re: [Asterisk-Users] 911 and ISDN PRI
I've talked to the carrier (verizon), what they said is that the call is not leavingmy phone equipment. I tried to tell him that I'm getting an error back from his system, but he insists that the channel never comes up. Their answers was talk to your telco vendor, its on their end. So I guess I'm pretty much SOL when it comes to using 911 with the PRI. Below is the debug, they wanted me to try all the DID numbers to see if it worked on any of them (40 numbers) and the billing number, wouldn't work with any of them. -- Executing SetCallerID(IAX2/sycam-16384, 8157548823) in new stack -- Executing Dial(IAX2/sycam-16384, Zap/g2/911) in new stack-- Making new call for cr 33385 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 617/0x269) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0c 41 80 38 31 35 37 35 34 38 38 32 33] Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '8157548823' ] [70 04 a1 39 31 31] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '911' ] -- Called g2/911 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 617/0x269) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 9c] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ]-- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangupNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/25-1' On 2/8/06, Watkins, Bradley [EMAIL PROTECTED] wrote: It looks like the outbound caller ID is not being set properly. Most of the carriers that I've dealt with will act exactly as you said if you do not set it to what is expected at the 911 center. In particular: Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3251' ] Your user number being sent is just the caller ID of the SIP channel. Regards, - Brad -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Joe Pukepail Sent: Tuesday, February 07, 2006 3:26 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 911 and ISDN PRI I have a call in with the carrier, below is the PRI debug, looks like it is getting hungup because of Invalid Number format, I did try to use Setcallerid to change the callerID to a DID number in a previous attempt, but it still didn't go through. Not sure if that invalid number format is the calling number or the number I'm calling. I'll let the list know the result. I would encourage everyone to test their 911 functionality (especially if you have a PRI), I almost didn't check it.The PRI is up and 411 works, so I almost assumed that 911 would work. Make sure you call the police station first to make sure they are not swamped by real emergency calls and let them know you are testing . -- Executing NoOp(SIP/3251-7316, 3251) in new stack -- Executing Dial(SIP/3251-7316, Zap/g2/911) in new stack-- Making new call for cr 33144 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=44 Call Ref: len= 2 (reference 376/0x178) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 09 b1 52 65 63 70 74 69 6f 6e] Display (len= 9) Charset: 31 [ Recption ] [6c 06 41 80 33 32 35 31] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user
RE: [Asterisk-Users] 911 and ISDN PRI
Joe, It is entirely possible, even probable, that you spoke with someone who doesnt know the difference between PRI and good ol fashion T1 trunks. If he insists that the channel never comes up then he is definitely looking in the wrong place. Assuming hes talking about the B channel, obviously its not coming up because thats what youre troubleshooting. If hes insisting that the D channel isnt coming up then obviously none of your calls would be working, DID or otherwise. Sounds like youve got a case of vendor wheel-of-blame going on. Please contact me off list and Ill be happy to help you out. I used to be a vendor so I know the routine. Ive got a dozen T1s, half of which are PRIs, from 3 different telcos so Im used to this kind of stuff. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail Sent: Wednesday, February 08, 2006 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 and ISDN PRI I've talked to the carrier (verizon), what they said is that the call is not leavingmy phone equipment. I tried to tell him that I'm getting an error back from his system, but he insists that the channel never comes up. Their answers was talk to your telco vendor, its on their end. So I guess I'm pretty much SOL when it comes to using 911 with the PRI. Below is the debug, they wanted me to try all the DID numbers to see if it worked on any of them (40 numbers) and the billing number, wouldn't work with any of them. -- Executing SetCallerID(IAX2/sycam-16384, 8157548823) in new stack -- Executing Dial(IAX2/sycam-16384, Zap/g2/911) in new stack -- Making new call for cr 33385 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 617/0x269) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0c 41 80 38 31 35 37 35 34 38 38 32 33] Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '8157548823' ] [70 04 a1 39 31 31] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '911' ] -- Called g2/911 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 617/0x269) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 9c] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/25-1' On 2/8/06, Watkins, Bradley [EMAIL PROTECTED] wrote: It looks like the outbound caller ID is not being set properly. Most of the carriers that I've dealt with will act exactly as you said if you do not set it to what is expected at the 911 center. In particular: Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3251' ] Your user number being sent is just the caller ID of the SIP channel. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Joe Pukepail Sent: Tuesday, February 07, 2006 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 and ISDN PRI I have a call in with the carrier, below is the PRI debug, looks like it is getting hungup because of Invalid Number format, I did try to use Setcallerid to change the callerID to a DID number in a previous attempt, but it still didn't go through. Not sure if that invalid number format is the calling number or the number I'm calling. I'll let the list know the result. I would encourage everyone to test their 911 functionality (especially if you have a PRI), I almost didn't check it.The PRI is up and 411 works, so I almost assumed that 911 would work. Make sure you call the police station first to make sure they are not swamped by real emergency calls and let them know you
Re: [Asterisk-Users] 911 and ISDN PRI
From me looking at it - it looks like the Telco is not accepting a 3 digit number. Have you tried 411 on the PRI to see if you are getting the same error? My 2 Cents -Jon Michael Collins wrote: Joe, It is entirely possible, even probable, that you spoke with someone who doesnt know the difference between PRI and good ol fashion T1 trunks. If he insists that the channel never comes up then he is definitely looking in the wrong place. Assuming hes talking about the B channel, obviously its not coming up because thats what youre troubleshooting. If hes insisting that the D channel isnt coming up then obviously none of your calls would be working, DID or otherwise. Sounds like youve got a case of vendor wheel-of-blame going on. Please contact me off list and Ill be happy to help you out. I used to be a vendor so I know the routine. Ive got a dozen T1s, half of which are PRIs, from 3 different telcos so Im used to this kind of stuff. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Joe Pukepail Sent: Wednesday, February 08, 2006 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 and ISDN PRI I've talked to the carrier (verizon), what they said is that the call is not leavingmy phone equipment. I tried to tell him that I'm getting an error back from his system, but he insists that the channel never comes up. Their answers was "talk to your telco vendor, its on their end". So I guess I'm pretty much SOL when it comes to using 911 with the PRI. Below is the debug, they wanted me to try all the DID numbers to see if it worked on any of them (40 numbers) and the billing number, wouldn't work with any of them. -- Executing SetCallerID("IAX2/sycam-16384", "8157548823") in new stack -- Executing Dial("IAX2/sycam-16384", "Zap/g2/911") in new stack -- Making new call for cr 33385 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 617/0x269) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0c 41 80 38 31 35 37 35 34 38 38 32 33] Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '8157548823' ] [70 04 a1 39 31 31] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '911' ] -- Called g2/911 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 617/0x269) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 9c] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/25-1' On 2/8/06, Watkins, Bradley [EMAIL PROTECTED] wrote: It looks like the outbound caller ID is not being set properly. Most of the carriers that I've dealt with will act exactly as you said if you do not set it to what is expected at the 911 center. In particular: Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3251' ] Your user number being sent is just the caller ID of the SIP channel. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Joe Pukepail Sent: Tuesday, February 07, 2006 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 and ISDN PRI I have a call in with the carrier, below is the PRI debug, looks like it is getting hungup because of "Invalid Number format", I did try to use Setcallerid to change the callerID to a DID number in a previous attempt, but it still didn't go through. Not sure if that "invalid number
Re: [Asterisk-Users] 911 and ISDN PRI
Joe, Verizon's tech is doing what most techs will do in a similar situation when they encounter an interop problem - blame the customer ;-) I've been involved in _MANY_ such calls, and you need to politely but firmly ask to be escalated to a switch engineer, or someone who has a good understanding of the ISDN protocol. You might want to ask them to 'run a trap on the call' as well ... that will sometimes impress them. You need them to dump the ISDN chatter between you and them for a single call (such as the one below) and they will need to get setup to collect that. Once they've dumped it, they will need someone fairly senior there to interpret it. You might not need to capture a call in this case though, since Asterisk provides you with all you need. Here's what the ISDN trace you include below tells you: 1. The PRI is in service - the D-channel and B-channel are working together to talk to your telco's switch 2. Their switch is rejecting your call with ISDN cause code 28 (Invalid number format). The called-party is clearly '911' in this call. Their switch seems to be rejecting that as an invalid number. You need to ask them why! ;-) -Darren - Original Message - From: Joe Pukepail To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, February 08, 2006 6:30 PM Subject: Re: [Asterisk-Users] 911 and ISDN PRI I've talked to the carrier (verizon), what they said is that the call is not leaving my phone equipment. I tried to tell him that I'm getting an error back from his system, but he insists that the channel never comes up. Their answers was talk to your telco vendor, its on their end. So I guess I'm pretty much SOL when it comes to using 911 with the PRI. Below is the debug, they wanted me to try all the DID numbers to see if it worked on any of them (40 numbers) and the billing number, wouldn't work with any of them. -- Executing SetCallerID(IAX2/sycam-16384, 8157548823) in new stack -- Executing Dial(IAX2/sycam-16384, Zap/g2/911) in new stack -- Making new call for cr 33385 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 617/0x269) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0c 41 80 38 31 35 37 35 34 38 38 32 33] Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '8157548823' ] [70 04 a1 39 31 31] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '911' ] -- Called g2/911 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 617/0x269) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 9c] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/25-1' On 2/8/06, Watkins, Bradley [EMAIL PROTECTED] wrote: It looks like the outbound caller ID is not being set properly. Most of the carriers that I've dealt with will act exactly as you said if you do not set it to what is expected at the 911 center. In particular: Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3251' ] Your user number being sent is just the caller ID of the SIP channel. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Joe Pukepail Sent: Tuesday, February 07, 2006 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 and ISDN PRI I have a call in with the carrier, below is the PRI debug, looks like it is getting hungup because of Invalid Number format, I did try to use
[Asterisk-Users] 911 and ISDN PRI
Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 911 and ISDN PRI
911 *should* work on a PRI. If you are getting a hangup and you dont see a valid hangupcause, it might be best to get your carrier on the line and have them monitor the circuit while you dial 911. They might be able to tell you what the problem is. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail Sent: Tuesday, February 07, 2006 10:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 911 and ISDN PRI Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 911 and ISDN PRI
I have used 911 with PRI with nothing else configured. Telco had to make changes to their router for DID numbers to call through. Adam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail Sent: Tuesday, February 07, 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 911 and ISDN PRI Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and ISDN PRI
I dunno about your provider but I know that 2 of my 3 MCI PRI circuits have no 911 abilities. MCI tells me this is becasue I have no local dialing plan on them. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Michael Collins wrote: 911 **should** work on a PRI. If you are getting a hangup and you don’t see a valid hangupcause, it might be best to get your carrier on the line and have them monitor the circuit while you dial 911. They might be able to tell you what the problem is. -MC *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Joe Pukepail *Sent:* Tuesday, February 07, 2006 10:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] 911 and ISDN PRI Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and ISDN PRI
I have a call in with the carrier, below is the PRI debug, looks like it is getting hungup because of Invalid Number format, I did try to use Setcallerid to change the callerID to a DID number in a previous attempt, but it still didn't go through. Not sure if that invalid number format is the calling number or the number I'm calling. I'll let the list know the result. I would encourage everyone to test their 911 functionality (especially if you have a PRI), I almost didn't check it.The PRI is up and 411 works, so I almost assumed that 911 would work. Make sure you call the police station first to make sure they are not swamped by real emergency calls and let them know you are testing . -- Executing NoOp(SIP/3251-7316, 3251) in new stack -- Executing Dial(SIP/3251-7316, Zap/g2/911) in new stack-- Making new call for cr 33144 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=44 Call Ref: len= 2 (reference 376/0x178) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 09 b1 52 65 63 70 74 69 6f 6e] Display (len= 9) Charset: 31 [ Recption ] [6c 06 41 80 33 32 35 31] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3251' ] [70 04 a1 39 31 31] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '911' ] -- Called g2/911 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 376/0x178) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 9c] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ]-- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangup On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote: I dunno about your provider but I know that 2 of my 3 MCI PRI circuitshave no 911 abilities. MCI tells me this is becasue I have no local dialing plan on them.Mark, G7LTT/KC2ENIRandolph, NJhttp://www.g7ltt.comMichael Collins wrote: 911 **should** work on a PRI.If you are getting a hangup and you don't see a valid hangupcause, it might be best to get your carrier on the line and have them monitor the circuit while you dial 911.They might be able to tell you what the problem is. -MC *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] *On Behalf Of *Joe Pukepail *Sent:* Tuesday, February 07, 2006 10:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] 911 and ISDN PRI Does asterisk support this?I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup.Does 911 normally work over a PRI line?Anything special I have to setup in asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 911 and ISDN PRI
Mark, It definitely sounds like the carrier is looking for something more than just 911 on the D channel. Please let us know what the carrier says about 911 dialing so that we can make sure our *s are all setup properly. Thanks, MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail Sent: Tuesday, February 07, 2006 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 and ISDN PRI I have a call in with the carrier, below is the PRI debug, looks like it is getting hungup because of Invalid Number format, I did try to use Setcallerid to change the callerID to a DID number in a previous attempt, but it still didn't go through. Not sure if that invalid number format is the calling number or the number I'm calling. I'll let the list know the result. I would encourage everyone to test their 911 functionality (especially if you have a PRI), I almost didn't check it.The PRI is up and 411 works, so I almost assumed that 911 would work. Make sure you call the police station first to make sure they are not swamped by real emergency calls and let them know you are testing . -- Executing NoOp(SIP/3251-7316, 3251) in new stack -- Executing Dial(SIP/3251-7316, Zap/g2/911) in new stack -- Making new call for cr 33144 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=44 Call Ref: len= 2 (reference 376/0x178) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 09 b1 52 65 63 70 74 69 6f 6e] Display (len= 9) Charset: 31 [ Recption ] [6c 06 41 80 33 32 35 31] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3251' ] [70 04 a1 39 31 31] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '911' ] -- Called g2/911 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 376/0x178) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 9c] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangup On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote: I dunno about your provider but I know that 2 of my 3 MCI PRI circuits have no 911 abilities. MCI tells me this is becasue I have no local dialing plan on them. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Michael Collins wrote: 911 **should** work on a PRI.If you are getting a hangup and you don't see a valid hangupcause, it might be best to get your carrier on the line and have them monitor the circuit while you dial 911.They might be able to tell you what the problem is. -MC *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] *On Behalf Of *Joe Pukepail *Sent:* Tuesday, February 07, 2006 10:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] 911 and ISDN PRI Does asterisk support this?I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup.Does 911 normally work over a PRI line?Anything special I have to setup in asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 911 Q
Currently we are working with Telco Providers to provide 911 and e911 with all the bells and whistles, including CNMAN features. This will enable you to deliver 911 calls to PSAP with out having to tell them your location. Get ready to manage DB ... check out REDSKY software -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joel Newkirk Sent: Monday, October 03, 2005 2:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 Q Thank you - while not directly an answer to my question, it directly addresses the root of my question, pointing me where I'll need to go to dig deeper. It also tells me what we didn't want to hear, that there's a very good possibility that we simply won't be able to ensure that the 911 call center can tell which unit a call comes from without verbal specification from the caller. j On Sun, 2005-10-02 at 08:13 -0600, Rich Adamson wrote: Asterisk is more then capable of sending the appropriate callerid info to any remote site including 911 centers. However, there is a telco between asterisk and the 911 center that may not have realistic policies/systems to accept and forward that callerid. So, your objective becomes one of what the telco will allow you to do (and their support of your objective). As one example only, the telco might have a switch that does not have PRI capabilities (I know of many of these), but they provide ANI info to the 911 centers since that _might_ be the only data they can provide. If that were the case in your environment, it doesn't make any difference what you do with asterisk, it won't be supported. I know from practical experience that a telco's switch (in most cases) will accept calleridnum via a PRI, but on most central office switches its an option that needs to be turned on. (Local telco policy _might_ say they will never do that.) Once that option is turned on, you can send almost anything to them in the form of calleridnum. The callerid name is a different story. The central office switch that _terminates_ any call (including 911 calls) will have a mechanism to do a database lookup/dip, and if that database has not been populated with an appropriate callerid name, will not provide callerid names to the 911 center (or anyone else). That essentially says you can program asterisk to send anything that you want from a callerid name perspective and it will be ignored in the US. In very general terms, only telco personnal have the access to update the callerid database, and usually that is limited to the CO prefixes they support. Also keep in mind that not all 911 centers are the same from a technical perspective. They certainly accept callerid numbers, but they may have their own local database for names (etc), or, they may also do a database lookup from some distant database. If you think about those customers that subscribe to callerid blocking, cell phones gps, and the requirements of 911 centers, its not hard to visualize several different 911 implementation approaches. Talk to a knowledgable telco person (might be somewhat difficult to find the appropriate person), and talk to the 911 center manager to better understand what options you might have available. I'd start with the 911 manager as he will know a telco person that understands the technical requirements. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Q
Thank you - while not directly an answer to my question, it directly addresses the root of my question, pointing me where I'll need to go to dig deeper. It also tells me what we didn't want to hear, that there's a very good possibility that we simply won't be able to ensure that the 911 call center can tell which unit a call comes from without verbal specification from the caller. j On Sun, 2005-10-02 at 08:13 -0600, Rich Adamson wrote: Asterisk is more then capable of sending the appropriate callerid info to any remote site including 911 centers. However, there is a telco between asterisk and the 911 center that may not have realistic policies/systems to accept and forward that callerid. So, your objective becomes one of what the telco will allow you to do (and their support of your objective). As one example only, the telco might have a switch that does not have PRI capabilities (I know of many of these), but they provide ANI info to the 911 centers since that _might_ be the only data they can provide. If that were the case in your environment, it doesn't make any difference what you do with asterisk, it won't be supported. I know from practical experience that a telco's switch (in most cases) will accept calleridnum via a PRI, but on most central office switches its an option that needs to be turned on. (Local telco policy _might_ say they will never do that.) Once that option is turned on, you can send almost anything to them in the form of calleridnum. The callerid name is a different story. The central office switch that _terminates_ any call (including 911 calls) will have a mechanism to do a database lookup/dip, and if that database has not been populated with an appropriate callerid name, will not provide callerid names to the 911 center (or anyone else). That essentially says you can program asterisk to send anything that you want from a callerid name perspective and it will be ignored in the US. In very general terms, only telco personnal have the access to update the callerid database, and usually that is limited to the CO prefixes they support. Also keep in mind that not all 911 centers are the same from a technical perspective. They certainly accept callerid numbers, but they may have their own local database for names (etc), or, they may also do a database lookup from some distant database. If you think about those customers that subscribe to callerid blocking, cell phones gps, and the requirements of 911 centers, its not hard to visualize several different 911 implementation approaches. Talk to a knowledgable telco person (might be somewhat difficult to find the appropriate person), and talk to the 911 center manager to better understand what options you might have available. I'd start with the 911 manager as he will know a telco person that understands the technical requirements. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Q
Joel Newkirk wrote on Friday, 30 September 2005 7:20 AM: Looking into setting up a couple asterisk servers at a country club, with VOIP phones in each of 100 short-term residential rental units. Approx 100 extensions, approx 24 outside lines. Since everything is geographically at one location, reaching 911 correctly shouldn't present a problem. However, the club wishes to ensure that 911 authorities are able to identify the precise rental unit placing the call. Mr. Newkirk, This and similar situations present a very serious issue for emergency responders. When you dial 911, your call is routed to the appropriate PSAP (Public Safety Answering Point) based on your ANI (Automatic Number Identification) or ELIN (Emergency Line Identification Number -- usually just another term for ANI). As your call arrives, the PSAP does a query of their ALI (Automatic Location Information) database to get your location information. Please note that the PSAP does NOT use Caller ID for this purpose. End users are not able to block their ANI (under normal circumstances), even though they may block their Caller ID. Either the ILEC or a company like Intrado will maintain the ALI database in your area. If you are getting your PRI and DIDs from your local ILEC, they would be responsible for getting the correct information entered into the ALI database. Typically, the information entered is only the physical address where the primary service is installed. In most circumstances, this information is enough to get police/fire/EMS to you in an emergency. However, I suspect the entire country club shares a single street address. If so, when someone dials 911, the PSAP will get only the main address of the country club. In this and similar situations, such as calling from within a multi-floor office building, a campus environment, etc., the main street address is simply not enough information to get emergency responders to you in a timely manner. Consider this not-so-unusual hypothetical scenario. A guest of the Pennsauken Country Club is having a heart attack in his bungalow. He dials 911. The dispatcher's screen at the PSAP shows the main information for the club (856) 662-4961 - 3800 Haddonfield Rd - Pennsauken Country Club - Pennsauken, NJ. The guest explains that he is experiencing severe chest pain, then either passes out before he can tell the dispatcher his exact location at the country club, or is confused or unaware of his exact location. The dispatcher would roll fire, EMS, and/or police to the main address. However, when they arrive, the emergency responders would have to knock on all 100+ doors to even attempt to determine who was having the emergency. Now you probably have a dead guest. Not good for business. First off, you should be using a PRI to connect your Asterisk server to the PSTN. You should also have a block of DIDs, with each guest room assigned its own, unique DID. This way you can differentiate among the individual rooms when people are making outbound calls, and guests may receive incoming calls in their room without going through an operator. Asterisk is capable of setting ANI in addition to Caller ID, on a per-call basis. This would ensure that the correct data is sent to the phone company when someone dials 911. As to getting the data to the PSAP to indicate where within the country club each DID is assigned, you have a couple of solutions. You can implement PS/ALI (Private Switch/Automatic Location Identification), or you can work with your telecom provider to have them enter the extended data into the ALI database for each DID individually. PS/ALI is the best solution, from a technical standpoint -- but it is usually quite expensive. PS/ALI allows you to provide the E-911 system with current, specific tenant location information to expedite emergency response times to the site of the emergency -- not just to the building or general site location. So when your guest having a heard attack in room 119 dials 911, the PSAP gets something more along the line of (856) 324-4119 - 3800 Haddonfield Rd - Building 5 Room 119 - Pennsauken Country Club - Pennsauken, NJ. PS/ALI is geared toward larger telecom users such as colleges, office buildings, large office campuses, etc., with a somewhat mobile population. It is utilized best when most of your extensions or DIDs are assigned to a person, as opposed to a location. This way, when the person moves from one office to another, your staff can push the change to the ALI database within minutes of the move, rather than phoning in a service order to the LEC, and waiting days for the change to be pushed to ALI. In your situation, I am assuming an extension or DID would most likely stay at a fixed location for quite some time (e.g. extension 4119 is always going to be guest room 119). So PS/ALI may be overkill in your situation. In that case, I would go the second route mentioned above. Work with your
Re: [Asterisk-Users] 911 Q
Think you might have jumped to a conclusion that might not be valid. If the telco can handle a PRI and will accept callerid from you, and each unit has a valid telephone number, then the telco can populate the callerid database with names. Those are the only two items the telco can provide in real time. The 911 center manager can better tell you exactly how they populate their database with street addresses and unit numbers. That process is likely an external non-automated process, or, the local telco is giving them the info via a electronic/paper copy of their service order. But, he's the only one that can tell you exactly how that works for his center. So, don't give. Go to the 911 manager and do some research; then go to his contact at the telco to get the real facts. Thank you - while not directly an answer to my question, it directly addresses the root of my question, pointing me where I'll need to go to dig deeper. It also tells me what we didn't want to hear, that there's a very good possibility that we simply won't be able to ensure that the 911 call center can tell which unit a call comes from without verbal specification from the caller. j On Sun, 2005-10-02 at 08:13 -0600, Rich Adamson wrote: Asterisk is more then capable of sending the appropriate callerid info to any remote site including 911 centers. However, there is a telco between asterisk and the 911 center that may not have realistic policies/systems to accept and forward that callerid. So, your objective becomes one of what the telco will allow you to do (and their support of your objective). As one example only, the telco might have a switch that does not have PRI capabilities (I know of many of these), but they provide ANI info to the 911 centers since that _might_ be the only data they can provide. If that were the case in your environment, it doesn't make any difference what you do with asterisk, it won't be supported. I know from practical experience that a telco's switch (in most cases) will accept calleridnum via a PRI, but on most central office switches its an option that needs to be turned on. (Local telco policy _might_ say they will never do that.) Once that option is turned on, you can send almost anything to them in the form of calleridnum. The callerid name is a different story. The central office switch that _terminates_ any call (including 911 calls) will have a mechanism to do a database lookup/dip, and if that database has not been populated with an appropriate callerid name, will not provide callerid names to the 911 center (or anyone else). That essentially says you can program asterisk to send anything that you want from a callerid name perspective and it will be ignored in the US. In very general terms, only telco personnal have the access to update the callerid database, and usually that is limited to the CO prefixes they support. Also keep in mind that not all 911 centers are the same from a technical perspective. They certainly accept callerid numbers, but they may have their own local database for names (etc), or, they may also do a database lookup from some distant database. If you think about those customers that subscribe to callerid blocking, cell phones gps, and the requirements of 911 centers, its not hard to visualize several different 911 implementation approaches. Talk to a knowledgable telco person (might be somewhat difficult to find the appropriate person), and talk to the 911 center manager to better understand what options you might have available. I'd start with the 911 manager as he will know a telco person that understands the technical requirements. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Q
On Monday 03 October 2005 12:17, Rich Adamson wrote: Think you might have jumped to a conclusion that might not be valid. If the telco can handle a PRI and will accept callerid from you, and each unit has a valid telephone number, then the telco can populate the callerid database with names. Those are the only two items the telco can provide in real time. I have some information from the 911 service manager for Bell Canada in Eastern Ontario. Basically the Public Service Automatic Location Indentification database (PSALI) only has allocations for BTNs (Billing Telephone Numbers) -- there are no ALI entries for DIDs from Bell Canada at this time, and there is no plan to do this. Basically if you set your outgoing ANI to a DID the PSAP office will have no address information, and indeed the switch may end up overwriting your ANI with the BTN. Since DIDs do not have an address associated with them (makes sense, they are only inward-numbers by design), you can convert DIDs to LDNs (Local Directory Number, same thing but has a directory (address) associated with it) -- the unfortunate side-effect of that is that LDNs are all billed separately so you would receive a separate bill for every LDN on a PRI. There is a service (of course!) being offered where you can provide specifically-formatted records for the PSALI database. It's not cheap, it's a $2000 setup fee and (IIRC) $500/mo for up to 500 record changes, and a two-year contract minimum. (These figures might be off, it's from memory.) However if you subscribe to this service you can assign any municipal address to any number and it will make its way into the PSALI database, which is what all the primary PSAP offices use to get the address information before routing the call to the appropriate secondary PSAP office. At least with Bell Canada, this is the only way to get your user's address information into the database used by the primary PSAP offices. The alternative, of course, is to set up your own primary PSAP system and then you can use whatever database and organization system you want, and redirect calls to the appropriate secondary PSAP office yourself. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Q
I installed a Marquee sign (aka reader board), which was sent emergency information via an RS-232 serial port. It was pretty nifty, as it was during to 'everywhere must have caller ID' phase in the 90s. Most signs are cheap, and can just be placed in the clubhouse window. You could even have nice littlearrows pointing the direction of the 911 caller's dwelling... Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Joel Newkirk [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, September 30, 2005 7:20 AM Subject: [Asterisk-Users] 911 Q OK, got a question on 911. Looking into setting up a couple asterisk servers at a country club, with VOIP phones in each of 100 short-term residential rental units. Approx 100 extensions, approx 24 outside lines. Since everything is geographically at one location, reaching 911 correctly shouldn't present a problem. However, the club wishes to ensure that 911 authorities are able to identify the precise rental unit placing the call. How can we achieve this, short of 'reciting' the unit number aloud at the beginning of the placed call? Thanks for any advice/tips. j ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Q
I installed a Marquee sign (aka reader board), which was sent emergency information via an RS-232 serial port. It was pretty nifty, as it was during to 'everywhere must have caller ID' phase in the 90s. Most signs are cheap, and can just be placed in the clubhouse window. You could even have nice littlearrows pointing the direction of the 911 caller's dwelling... Chris Coulthurst [EMAIL PROTECTED] - Original Message - OK, got a question on 911. Looking into setting up a couple asterisk servers at a country club, with VOIP phones in each of 100 short-term residential rental units. Approx 100 extensions, approx 24 outside lines. Since everything is geographically at one location, reaching 911 correctly shouldn't present a problem. However, the club wishes to ensure that 911 authorities are able to identify the precise rental unit placing the call. How can we achieve this, short of 'reciting' the unit number aloud at the beginning of the placed call? Thanks for any advice/tips. Asterisk is more then capable of sending the appropriate callerid info to any remote site including 911 centers. However, there is a telco between asterisk and the 911 center that may not have realistic policies/systems to accept and forward that callerid. So, your objective becomes one of what the telco will allow you to do (and their support of your objective). As one example only, the telco might have a switch that does not have PRI capabilities (I know of many of these), but they provide ANI info to the 911 centers since that _might_ be the only data they can provide. If that were the case in your environment, it doesn't make any difference what you do with asterisk, it won't be supported. I know from practical experience that a telco's switch (in most cases) will accept calleridnum via a PRI, but on most central office switches its an option that needs to be turned on. (Local telco policy _might_ say they will never do that.) Once that option is turned on, you can send almost anything to them in the form of calleridnum. The callerid name is a different story. The central office switch that _terminates_ any call (including 911 calls) will have a mechanism to do a database lookup/dip, and if that database has not been populated with an appropriate callerid name, will not provide callerid names to the 911 center (or anyone else). That essentially says you can program asterisk to send anything that you want from a callerid name perspective and it will be ignored in the US. In very general terms, only telco personnal have the access to update the callerid database, and usually that is limited to the CO prefixes they support. Also keep in mind that not all 911 centers are the same from a technical perspective. They certainly accept callerid numbers, but they may have their own local database for names (etc), or, they may also do a database lookup from some distant database. If you think about those customers that subscribe to callerid blocking, cell phones gps, and the requirements of 911 centers, its not hard to visualize several different 911 implementation approaches. Talk to a knowledgable telco person (might be somewhat difficult to find the appropriate person), and talk to the 911 center manager to better understand what options you might have available. I'd start with the 911 manager as he will know a telco person that understands the technical requirements. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Q
OK, got a question on 911. Looking into setting up a couple asterisk servers at a country club, with VOIP phones in each of 100 short-term residential rental units. Approx 100 extensions, approx 24 outside lines. Since everything is geographically at one location, reaching 911 correctly shouldn't present a problem. However, the club wishes to ensure that 911 authorities are able to identify the precise rental unit placing the call. How can we achieve this, short of 'reciting' the unit number aloud at the beginning of the placed call? Thanks for any advice/tips. j ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Q
With hotel systems When some places a 911 call it is printed on the printer in the Front Desk, Hwen help arrives they usually go to the Frount Dsek anyway. I would set up a System() that would not only printout he romm number on the Front Desk Printer but also drop a call file in to trigger a call to the Front Desk with a prerecorded message of wht extention just called 911. That way the Hotel can send someone to the room to act as first response and the Frount Desk can direct the 911 team to the correct location. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joel Newkirk Sent: Friday, September 30, 2005 10:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 911 Q OK, got a question on 911. Looking into setting up a couple asterisk servers at a country club, with VOIP phones in each of 100 short-term residential rental units. Approx 100 extensions, approx 24 outside lines. Since everything is geographically at one location, reaching 911 correctly shouldn't present a problem. However, the club wishes to ensure that 911 authorities are able to identify the precise rental unit placing the call. How can we achieve this, short of 'reciting' the unit number aloud at the beginning of the placed call? Thanks for any advice/tips. j ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Q
On Fri, Sep 30, 2005 at 10:20:12AM -0400, Joel Newkirk wrote: How can we achieve this, short of 'reciting' the unit number aloud at the beginning of the placed call? Hmm, could you just put the full address (including unit no.) in the E911 database for the corresponding numbers assigned? You might have to work with your phone company/LEC on this, but I think it would be the most transparent solution. Ray ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Q
On Fri, 2005-09-30 at 09:16 -0700, Ray Van Dolson wrote: On Fri, Sep 30, 2005 at 10:20:12AM -0400, Joel Newkirk wrote: How can we achieve this, short of 'reciting' the unit number aloud at the beginning of the placed call? Hmm, could you just put the full address (including unit no.) in the E911 database for the corresponding numbers assigned? We're only expecting about 1/4 as many outside lines as rental units, which I believe would make that impossible. These units are typically rented by golfers and hangers-on during various events. (think about the cloud that descends on a US Open golf tournament) It's not hotel-style (there's no 'front desk') nor long-term residences - it's a condominium complex on the course dedicated entirely to short-term rentals. The situation is that someone calling in will be prompted for a unit number, outbound calling 'appears' direct dialed. j You might have to work with your phone company/LEC on this, but I think it would be the most transparent solution. Ray ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Notices
Packet8 got around this in an interesting waycharge clients $1.50 per month for E911 or have the option of saying no. Lol, how many people do you think took them up on that offer? Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Friday, 26 August 2005 6:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 Notices On 8/26/05, Mark Phillips [EMAIL PROTECTED] wrote: Broadvoice sent out a notice and threatened to disconnect me if I did not respond. If I disagreed with their stand they would disconnect me too. I think they said something like we don't have it and we ain't getting it. Click here to acknowledge. I'm guessing that the statement gets them off the hook? The way I understand it. Yes, for now. That only allows them to be compliant up until the mandatory compliance date. After that date passes, technically, you're supposed to offer it if you're business is interconnecting voip networks to the PSTN. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Notices
Remarks inline Dean Collins wrote: Packet8 got around this in an interesting waycharge clients $1.50 per month for E911 or have the option of saying no. Lol, how many people do you think took them up on that offer? From what I understand, Packet8 had this option for quite some time. I used (more than one year ago) to be Packet8 customer. I still use a couple of DTA310 in my * system :-) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Friday, 26 August 2005 6:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 Notices On 8/26/05, Mark Phillips [EMAIL PROTECTED] wrote: Broadvoice sent out a notice and threatened to disconnect me if I did not respond. If I disagreed with their stand they would disconnect me too. I think they said something like we don't have it and we ain't getting it. Click here to acknowledge. I'm guessing that the statement gets them off the hook? The way I understand it. Yes, for now. That only allows them to be compliant up until the mandatory compliance date. After that date passes, technically, you're supposed to offer it if you're business is interconnecting voip networks to the PSTN. ___ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Notices
With the deadline coming up for sending notices to customers, I found it curious that out of 4-5 different providers I use, to date only one of them has contacted me. The rest don't even have anything on their website that I could find. Junction Networks was the only one that actually sent me a letter and also have everything right on the first page when you login to their system. A week or so ago I remember reading an article where the CEO from one of my vendors was complaining that they wouldn't have enough time to get all of their customers to respond in time. I thought that was pretty funny given that they don't seem to even be contacting anyone yet. There isn't even anything on their website except a statement that they do not plan to support 911 anytime soon. Am I missing something here? Is the FCC going to be extending deadlines and that's why the apparent lack of action on this issue? Just curious. I thought I would have started receiving letters a long time ago. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Notices
An extension of 30 days has been granted. Just like the HDTV broadcast requirement deadlines the FCC cooked up I'd predict there will be a few more extensions before the fight is over. http://tinyurl.com/a8tj8 -- Reuters article Am I missing something here? Is the FCC going to be extending deadlines and that's why the apparent lack of action on this issue? Just curious. I thought I would have started receiving letters a long time ago. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Notices
Broadvoice sent out a notice and threatened to disconnect me if I did not respond. If I disagreed with their stand they would disconnect me too. I think they said something like we don't have it and we ain't getting it. Click here to acknowledge. I'm guessing that the statement gets them off the hook? snacktime wrote: With the deadline coming up for sending notices to customers, I found it curious that out of 4-5 different providers I use, to date only one of them has contacted me. The rest don't even have anything on their website that I could find. Junction Networks was the only one that actually sent me a letter and also have everything right on the first page when you login to their system. A week or so ago I remember reading an article where the CEO from one of my vendors was complaining that they wouldn't have enough time to get all of their customers to respond in time. I thought that was pretty funny given that they don't seem to even be contacting anyone yet. There isn't even anything on their website except a statement that they do not plan to support 911 anytime soon. Am I missing something here? Is the FCC going to be extending deadlines and that's why the apparent lack of action on this issue? Just curious. I thought I would have started receiving letters a long time ago. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Notices
On 8/26/05, Mark Phillips [EMAIL PROTECTED] wrote: Broadvoice sent out a notice and threatened to disconnect me if I did not respond. If I disagreed with their stand they would disconnect me too. I think they said something like we don't have it and we ain't getting it. Click here to acknowledge. I'm guessing that the statement gets them off the hook? The way I understand it. Yes, for now. That only allows them to be compliant up until the mandatory compliance date. After that date passes, technically, you're supposed to offer it if you're business is interconnecting voip networks to the PSTN. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Notices
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Friday, August 26, 2005 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 Notices After that date passes, technically, you're supposed to offer it if you're business is interconnecting voip networks to the PSTN. ___ An important distinction should be clarified here. The FCC will require interconnected VOIP providers to provide 911/E911 service. Those VOIP operators providing dial-tone and a DID number need to comply. Those providing termination, or origination alone will not. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.15/82 - Release Date: 08/25/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Notices
Uh how's that work? FCC requires voip providers to offer it... how on earth is broadvoice NOT offering it and being legal? On 8/26/05, Mark Phillips [EMAIL PROTECTED] wrote: Broadvoice sent out a notice and threatened to disconnect me if I did not respond. If I disagreed with their stand they would disconnect me too. I think they said something like we don't have it and we ain't getting it. Click here to acknowledge. I'm guessing that the statement gets them off the hook? snacktime wrote: With the deadline coming up for sending notices to customers, I found it curious that out of 4-5 different providers I use, to date only one of them has contacted me. The rest don't even have anything on their website that I could find. Junction Networks was the only one that actually sent me a letter and also have everything right on the first page when you login to their system. A week or so ago I remember reading an article where the CEO from one of my vendors was complaining that they wouldn't have enough time to get all of their customers to respond in time. I thought that was pretty funny given that they don't seem to even be contacting anyone yet. There isn't even anything on their website except a statement that they do not plan to support 911 anytime soon. Am I missing something here? Is the FCC going to be extending deadlines and that's why the apparent lack of action on this issue? Just curious. I thought I would have started receiving letters a long time ago. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Service Providers
Who is everyone contracting with for 911 services with the upcoming FCC deadline? I've got a few feelers out there working on this issue, but no real solid leads yet. ___ 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] 911 Service Providers
We have done it with Group Telecom in Canada but we have to ask customer to keep their ATAs at fixed place. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Oster Sent: Monday, July 25, 2005 2:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 911 Service Providers Who is everyone contracting with for 911 services with the upcoming FCC deadline? I've got a few feelers out there working on this issue, but no real solid leads yet. ___ 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
Re: [Asterisk-Users] 911 context, is this right?
On Friday 03 June 2005 05:50, Chris Coulthurst wrote: I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten = 911,1,ChanIsAvail(Zap/1) exten = 911,2,Dial(Zap/1/911) exten = 911,3,Hangup() exten = 911,102,ChanIsAvail(Zap/4) exten = 911,103,Dial(Zap/4/911) exten = 911,104,Hangup() exten = 911,203,ChanIsAvail(Zap/5) exten = 911,204,Dial(Zap/5/911) exten = 911,205,Hangup() exten = 911,304,SoftHangup(Zap/5-1) exten = 911,305,Wait(2) exten = 911,306,Goto(204) Why would you do this? Use a group: zaptel.conf: group = 9 channel = 1,4,5 [e911] exten = 911,1,Dial(Zap/g9/ww911) exten = 911,102,SoftHangup(Zap/5) exten = 911,103,Goto(1) Basically dial using the first free line in group 9. If the Dial fails, hang up zap/5 and try again. I added two 'w's in the dial string just to make sure the telco switch is ready to receive DTMF (this may not be necessary) I'm not checking other lines than 5 (there's an assumption that line 5 is always going to work but in an emergency situation I'd just as soon soft hangup all 3 channels and try again. I also *TOTALLY* disagree with using Ringing() to calm the caller. If the call's not going through they SHOULD be thinking of using an alternative way to reach 911, not calmly waiting for an answer that just wont come. -A. ___ 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] 911 context, is this right?
On Mon, 2005-06-06 at 07:17 -0400, Andrew Kohlsmith wrote: On Friday 03 June 2005 05:50, Chris Coulthurst wrote: I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? Why would you do this? Use a group: Yes, use a group... but... zaptel.conf: group = 9 channel = 1,4,5 [e911] exten = 911,1,Dial(Zap/g9/ww911) exten = 911,102,SoftHangup(Zap/5) exten = 911,103,Goto(1) Boom, you just hungup on the emergency call that was already in progress I wouldn't call that even close to ideal. PS, you might at least somehow randomise the line you will hangup on... Basically dial using the first free line in group 9. If the Dial fails, hang up zap/5 and try again. I added two 'w's in the dial string just to make sure the telco switch is ready to receive DTMF (this may not be necessary) This will delay the call being sent, with absolutely no feedback to the caller. I'm not checking other lines than 5 (there's an assumption that line 5 is always going to work but in an emergency situation I'd just as soon soft hangup all 3 channels and try again. Yes, line 5 may not work, and also line 5 is more likely to have another emergency call in progress. I disagree with hanging up channels in this manner... IMHO, it is worse to hangup an emergency call in-progress than to simply return congestion You must check that the call in progress isn't itself an emergency call. I also *TOTALLY* disagree with using Ringing() to calm the caller. If the call's not going through they SHOULD be thinking of using an alternative way to reach 911, not calmly waiting for an answer that just wont come. But the call *IS* going through, I just allowed the caller to hear ringing for 2 seconds instead of dead-air. We just made a line available for him, so what makes you think it won't go through? (OK, someone else might steal the line while we are waiting...). Of course, after the two seconds, if the line is busy, they will hear busy, and then be able to decide the best course of action Perhaps retry, etc... In any case, whether you use a group, or play ringing, or don't, etc... IMHO, is irrelevant, what all of these dialplans are missing is the importance of NOT disconnecting an emergency call which is in-progress. Of course, that is just my 0.02c worth... Regards, Adam ___ 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] 911 context, is this right?
I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? Why would you do this? Use a group: Yes, use a group... but... zaptel.conf: group = 9 channel = 1,4,5 [e911] exten = 911,1,Dial(Zap/g9/ww911) exten = 911,102,SoftHangup(Zap/5) exten = 911,103,Goto(1) Boom, you just hungup on the emergency call that was already in progress I wouldn't call that even close to ideal. PS, you might at least somehow randomise the line you will hangup on... Basically dial using the first free line in group 9. If the Dial fails, hang up zap/5 and try again. I added two 'w's in the dial string just to make sure the telco switch is ready to receive DTMF (this may not be necessary) This will delay the call being sent, with absolutely no feedback to the caller. I'm not checking other lines than 5 (there's an assumption that line 5 is always going to work but in an emergency situation I'd just as soon soft hangup all 3 channels and try again. Yes, line 5 may not work, and also line 5 is more likely to have another emergency call in progress. I disagree with hanging up channels in this manner... IMHO, it is worse to hangup an emergency call in-progress than to simply return congestion You must check that the call in progress isn't itself an emergency call. I also *TOTALLY* disagree with using Ringing() to calm the caller. If the call's not going through they SHOULD be thinking of using an alternative way to reach 911, not calmly waiting for an answer that just wont come. But the call *IS* going through, I just allowed the caller to hear ringing for 2 seconds instead of dead-air. We just made a line available for him, so what makes you think it won't go through? (OK, someone else might steal the line while we are waiting...). Of course, after the two seconds, if the line is busy, they will hear busy, and then be able to decide the best course of action Perhaps retry, etc... In any case, whether you use a group, or play ringing, or don't, etc... IMHO, is irrelevant, what all of these dialplans are missing is the importance of NOT disconnecting an emergency call which is in-progress. Of course, that is just my 0.02c worth... Never did answer whether this effort is focused on a home system or on a small business. It does make a difference. If you are insistent on doing the above, then at least consider giving the second 911 caller a recorded message that says a 911 call is in progress instead of arbitrarily dumping _any_ calls. Assuming a reasonable size fire in a business, you're almost guaranteed to have multiple 911 calls originating from employees that don't have a clue that other calls are already in progress. By using the call dumping approach, you couldn't possibly program a resonable dialplan that takes every assumption into consideration, regardless of how you program it. Murphy's law also says your system/dialplan will fail at the most inopportune time. Therefore, if you don't have an alternative plan (that does not rely on *), we'll watch for your case to show up in the court records. /* end of comments on this thread for me */ ___ 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] 911 context, is this right?
On Fri, 2005-06-03 at 08:28 -0600, Rich Adamson wrote: I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten = 911,1,ChanIsAvail(Zap/1) exten = 911,2,Dial(Zap/1/911) exten = 911,3,Hangup() exten = 911,102,ChanIsAvail(Zap/4) exten = 911,103,Dial(Zap/4/911) exten = 911,104,Hangup() exten = 911,203,ChanIsAvail(Zap/5) exten = 911,204,Dial(Zap/5/911) exten = 911,205,Hangup() exten = 911,304,SoftHangup(Zap/5-1) exten = 911,305,Wait(2) exten = 911,306,Goto(204) Did I get the Priority + 101 idea right here? In your example above, if ChanIsAvail(Zap/1) finds that Zap/1 is unavailable, then priority 102 will be executed. However, what do you want to happen if Zap/1 is available (at least from asterisk's perspective), but the pstn line on Zap/1 doesn't process the call for whatever reason? One thing that bothers me with this style of dialplan for emergency numbers... What happens if you already have a emergency call on zap/5 ?? You have disconnected one call to emergency in order to place a second call, IMHO, that isn't a good thing to do :) So, question is, how can you ID that zap/5 is *currently* involved in a emergency call? exten = 911,1,ChanIsAvail(Zap/1) exten = 911,2,setgroup(911zap1) exten = 911,3,Dial(Zap/1/911) exten = 911,4,hangup exten = 911,102,setgroup(911zap1) exten = 911,103,checkgroup(1) exten = 911,104,softhangup(Zap/1-1) exten = 911,105,ringing ; Otherwise user might give up waiting and panic exten = 911,106,wait(2) exten = 911,107,goto(1) exten = 911,203,congestion Anyone like to comment on something like that?? Obviously, it can be extended to handle multiple lines, etc... and it should avoid dis-connecting an existing 911 call. Note, you should check priority numbers, and etc... this is just off the top of my head... Regards, Adam ___ 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] 911 context, is this right?
I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten = 911,1,ChanIsAvail(Zap/1) exten = 911,2,Dial(Zap/1/911) exten = 911,3,Hangup() exten = 911,102,ChanIsAvail(Zap/4) exten = 911,103,Dial(Zap/4/911) exten = 911,104,Hangup() exten = 911,203,ChanIsAvail(Zap/5) exten = 911,204,Dial(Zap/5/911) exten = 911,205,Hangup() exten = 911,304,SoftHangup(Zap/5-1) exten = 911,305,Wait(2) exten = 911,306,Goto(204) Did I get the Priority + 101 idea right here? In your example above, if ChanIsAvail(Zap/1) finds that Zap/1 is unavailable, then priority 102 will be executed. However, what do you want to happen if Zap/1 is available (at least from asterisk's perspective), but the pstn line on Zap/1 doesn't process the call for whatever reason? One thing that bothers me with this style of dialplan for emergency numbers... What happens if you already have a emergency call on zap/5 ?? You have disconnected one call to emergency in order to place a second call, IMHO, that isn't a good thing to do :) So, question is, how can you ID that zap/5 is *currently* involved in a emergency call? exten = 911,1,ChanIsAvail(Zap/1) exten = 911,2,setgroup(911zap1) exten = 911,3,Dial(Zap/1/911) exten = 911,4,hangup exten = 911,102,setgroup(911zap1) exten = 911,103,checkgroup(1) exten = 911,104,softhangup(Zap/1-1) exten = 911,105,ringing ; Otherwise user might give up waiting and panic exten = 911,106,wait(2) exten = 911,107,goto(1) exten = 911,203,congestion Anyone like to comment on something like that?? Obviously, it can be extended to handle multiple lines, etc... and it should avoid dis-connecting an existing 911 call. Note, you should check priority numbers, and etc... this is just off the top of my head... That's all getting pretty messy, but you could stuff a value using DBput when a 911 call is made, and clear the value at the end of that call. Then on subsequent calls, use DBget to see if a 911 call is in progress. But as mentioned previously, you've made a large number of assumptions to even think you can place a 911 call this way. You could probably do something like that with an AGI script, and have a fair amount more intelligence in the decision tree. If I recall correctly, seems like someone was taking about adding global variables into asterisk, but I don't recall how global those might be (if they are even there right now). That might also be a way to track number of outstanding 911 calls. ___ 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] 911 context, is this right?
On Sun, 2005-06-05 at 21:13 -0600, Rich Adamson wrote: exten = 911,1,ChanIsAvail(Zap/1) exten = 911,2,setgroup(911zap1) exten = 911,3,Dial(Zap/1/911) exten = 911,4,hangup exten = 911,102,setgroup(911zap1) exten = 911,103,checkgroup(1) exten = 911,104,softhangup(Zap/1-1) exten = 911,105,ringing ; Otherwise user might give up waiting and panic exten = 911,106,wait(2) exten = 911,107,goto(1) exten = 911,203,congestion That's all getting pretty messy, but you could stuff a value using DBput when a 911 call is made, and clear the value at the end of that call. Then on subsequent calls, use DBget to see if a 911 call is in progress. What is better about dbput/dbget compared to setgroup/checkgroup ?? Also, how do you ensure that the db value will be removed at hangup time? setgroup/checkgroup do this automatically for you ... But as mentioned previously, you've made a large number of assumptions to even think you can place a 911 call this way. Like what? (Seriously...) You could probably do something like that with an AGI script, and have a fair amount more intelligence in the decision tree. Of course, but there is a lot that you *can* do in an AGI that you *can* do in the dialplan... It might look better/cleaner as an AGI, but the dialplan would probably have lower overhead Also, do you want to rely on some external scripting language for emergency calls? What if someone upgraded your version of (for example) perl, so that now your AGI doesn't work?? or some module has been removed etc... If I recall correctly, seems like someone was taking about adding global variables into asterisk, but I don't recall how global those might be (if they are even there right now). That might also be a way to track number of outstanding 911 calls. Again, how do you remove that global variable ?? I thought about those two option before discarding them and proposing the above If you could let me know why you think global variables or db values are 'better' I'd appreciate it. Regards, Adam ___ 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] 911 context, is this right?
I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten = 911,1,ChanIsAvail(Zap/1) exten = 911,2,Dial(Zap/1/911) exten = 911,3,Hangup() exten = 911,102,ChanIsAvail(Zap/4) exten = 911,103,Dial(Zap/4/911) exten = 911,104,Hangup() exten = 911,203,ChanIsAvail(Zap/5) exten = 911,204,Dial(Zap/5/911) exten = 911,205,Hangup() exten = 911,304,SoftHangup(Zap/5-1) exten = 911,305,Wait(2) exten = 911,306,Goto(204) Did I get the Priority + 101 idea right here? Chris Coulthurst [EMAIL PROTECTED] ___ 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] 911 context, is this right?
I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten = 911,1,ChanIsAvail(Zap/1) exten = 911,2,Dial(Zap/1/911) exten = 911,3,Hangup() exten = 911,102,ChanIsAvail(Zap/4) exten = 911,103,Dial(Zap/4/911) exten = 911,104,Hangup() exten = 911,203,ChanIsAvail(Zap/5) exten = 911,204,Dial(Zap/5/911) exten = 911,205,Hangup() exten = 911,304,SoftHangup(Zap/5-1) exten = 911,305,Wait(2) exten = 911,306,Goto(204) Did I get the Priority + 101 idea right here? Probably a better way to do that is to use group=17 within the zapata.conf definitions for zap/1, zap/4, and zap/5. Then use something like exten = 911,1,Dial(Zap/g17/${EXTEN}) In your example above, if ChanIsAvail(Zap/1) finds that Zap/1 is unavailable, then priority 102 will be executed. However, what do you want to happen if Zap/1 is available (at least from asterisk's perspective), but the pstn line on Zap/1 doesn't process the call for whatever reason? If you're doing the above for a home system, you can probably handle failures in lots of different ways. But, if you're doing this for a business client, the above approach will leave you open for a fair number of legal liability issues when it doesn't work as expected. Just as a simple test, disconnect the pstn line from Zap/1 and see what happens when a call is placed. ___ 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] 911 context, is this right?
If Zap/5 is the least-used line, dial that one first :) Other than that, you could use a dial-group as someone else suggested. -Original Message- From: Chris Coulthurst [mailto:[EMAIL PROTECTED] Sent: Friday, June 03, 2005 4:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 911 context, is this right? I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten = 911,1,ChanIsAvail(Zap/1) exten = 911,2,Dial(Zap/1/911) exten = 911,3,Hangup() exten = 911,102,ChanIsAvail(Zap/4) exten = 911,103,Dial(Zap/4/911) exten = 911,104,Hangup() exten = 911,203,ChanIsAvail(Zap/5) exten = 911,204,Dial(Zap/5/911) exten = 911,205,Hangup() exten = 911,304,SoftHangup(Zap/5-1) exten = 911,305,Wait(2) exten = 911,306,Goto(204) Did I get the Priority + 101 idea right here? Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-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
[Asterisk-Users] 911 Options
I am curious if anybody has pointers on the best way to get the 7 digit PSAP number for an area. I am thinking about making a '911' extension that will dial the PSAP number, wait for the PSAP to answer and play a message giving the address of the originating call, and replay the the information every three minutes. I am concerned what may happen if my children try to dial 911 in an emergency but do not yet know our address. How are other people handling this? ___ 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] 911 Options
On Sun, 2005-05-15 at 15:55 -0600, Ira Burton wrote: I am curious if anybody has pointers on the best way to get the 7 digit PSAP number for an area. I am thinking about making a '911' extension that will dial the PSAP number, wait for the PSAP to answer and play a message giving the address of the originating call, and replay the the information every three minutes. I am concerned what may happen if my children try to dial 911 in an emergency but do not yet know our address. You can buy them on CD, however to do E911 you have to have a special trunk to the switch that the PSAP is off of, which transmits the E parts of E911 not just the audio. Where to buy them I dont know offhand, I do specifically recall seeing pages that sold national CDs (how adt, onstar, even other PSAPs contact a specific PSAP when needed). I do remember that I was googling psap administrators and other such things. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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