Re: [Asterisk-Users] What is Multi-layered-Access control
Many PBX's based on the Asterisk code refer to this in their feature list. Its taken off of the Asterisk.org site. Where is this used in Asterisk? Is it in the extensions.conf, is it used when logging into the console, where? How is it used? Is this referring top contexts in the dial plan? Message: 1 Date: Thu, 13 Apr 2006 23:30:24 -0400 From: Rusty Dekema [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] What is Multi-layerAccess control To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Well, I guess that would depend on what you mean by multi-layered access control. Kind of like your subject line asks. -Rusty On 4/13/06, Carey Mould [EMAIL PROTECTED] wrote: How does multi-layered access control work 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
[Asterisk-Users] Dial command exits non-zero
Why does my dial command exit non-zero when the calling party hangs up?
I am using a t1 with the following configuration:
/*from the zaptel.conf */
pan=1,1,0,esf,b8zs
em=1-24
=
/*from zapata.conf*/
[channels]
language=en
signalling=em_w
; change signalling to featd when telco sends callerid and DID after
group=1
context=ijt1
channel = 1-24dd;===dd;=
===
*/from extensions.conf*/
[ijt1]
exten = _X.,1,DISA(no-password|voip-cc)
[voip-cc]
exten = _X.,1,SetVar,milsecBal=5+0
exten = _X.,2,ResetCDR()
exten = _X.,3,SetCDRUserField(5599)
exten = _X.,4,NoOp(before dial)
exten = _X.,5,NoOp(before dial)
exten = _X.,6,Dial(IAX2/qcslink-iax2/${EXTEN},80,L(${milsecBal}:3))
exten = _X.,7,NoOp(${ANSWEREDTIME})
exten = _X.,8,NoOp(after dial)
===
the priority items after the dial is never reached if the caller
terminates the call. The dial exits non-zero... If I add the g
option in the dial the priority items continue
___
--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] continuing the context after caller hangsup
Hope the team can answer this: I have an asterisk setup that accepts calls from a T1 using the DISA command as follows: [emoclew-voip] exten = 9204883,1,DISA(no-password|voip-cc) I provide a dial tone using DISA, and validate against issued PINS using mysql command. Once validated the calls are placed to my provider. The problem I am having is that if the caller hangsup the context that DISA passes the call to (voip-cc) terminates. I am doing post call processing after the hangup, but none of the priorities after the dial executes. I have used the g option in the dial command but that only works when the called party hangsup. I have read all that is on the wiki about the special h extension but based on the comments it seems that I should stay away from it. My asterisk installation is using 1.09 running on Mandrake Linux 9.1. ___ --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] sip phone extensions at a remote site
I am in the proscess of integrating a clients remote and head office phone systems. Currenty each office has their own PBX and trunk lines. I am recommending that they put in an Asterisk server at the Head office with a WAN link to the remote office and switch to IP phones. Trunk lines at the remote site would be returned to the TELCO. External calls over the PSTN from the remote office would be routed over the WAN to the head office and through Asterisk to the PSTN trunk lines. All phones would then become extensions (both remote and head office locations). I want Person A in the remote office to dial an extension number and get Person B in the head office. What I am unsure about is if person A and Person B are both at the remote site and Asterisk PBX is at the head office, can A and B talk directly to each pther without traversing the WAN link? Has anyone done this before? What is the quality of the call if they have? Any information is useful. begin:vcard fn:Carey Mould n:Mould;Carey org:E2 Systems Limited adr:237 Old hope Road;;Suite 11 12, technology Innovation Centre;Kingston;;Kgn 6;Jamaica email;internet:[EMAIL PROTECTED] title:CEO/Consultant tel;work:(876) 512-2680 x-mozilla-html:FALSE url:http://www.e2team.com version:2.1 end:vcard ___ 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] Can a SIP Phone talk directly with anoyher SIP phone (ext a to Ext b)
I am in the proscess of integrating a clients remote and head office phone systems. Currenty each office has their own PBX and trunk lines. I am recommending that they put in an Asterisk server at the Head office with a WAN link to the remote office and switch to IP phones. Trunk lines at the remote site would be returned to the TELCO. External calls over the PSTN from the remote office would be routed over the WAN to the head office and through Asterisk to the PSTN trunk lines. All phones would then become extensions (both remote and head office locations). I want Person A in the remote office to dial an extension number and get Person B in the head office. What I am unsure about is if person A and Person B are both at the remote site and Asterisk PBX is at the head office, can A and B talk directly to each pther without traversing the WAN link? Has anyone done this before? What is the quality of the call if they have? Any information is useful. begin:vcard fn:Carey Mould n:Mould;Carey org:E2 Systems Limited adr:237 Old hope Road;;Suite 11 12, technology Innovation Centre;Kingston;;Kgn 6;Jamaica email;internet:[EMAIL PROTECTED] title:CEO/Consultant tel;work:(876) 512-2680 x-mozilla-html:FALSE url:http://www.e2team.com version:2.1 end:vcard ___ 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] RE: Sangoma VS. Digium
Where is this discussion going. I am about to do an installation that will require t1 interfaces. I am new to the telephone world and found the original discussion useful. I need to know from a reliability and performance standpoint what is the better choice. Sangoma or Digium? begin:vcard fn:Carey Mould n:Mould;Carey org:E2 Systems Limited adr:237 Old hope Road;;Suite 11 12, technology Innovation Centre;Kingston;;Kgn 6;Jamaica email;internet:[EMAIL PROTECTED] title:CEO/Consultant tel;work:(876) 512-2680 x-mozilla-html:FALSE url:http://www.e2team.com version:2.1 end:vcard ___ 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] Recommended GSM gateway
My client is looking fro a GSM gateway (24 ports). Any recommendations from the list. Anyone hase experience with the Orion GSM gateway? Also any experience integrating with a Meridan Option 51c PBX. Dont want to reinvent the wheel. I also thought I might put asterisk between the Meridian and the GSM gateway to provide authentication and call accounting, or even call quota's. You coments appreciated. begin:vcard fn:Carey Mould n:Mould;Carey org:E2 Systems Limited adr:237 Old hope Road;;Suite 11 12, technology Innovation Centre;Kingston;;Kgn 6;Jamaica email;internet:[EMAIL PROTECTED] title:CEO/Consultant tel;work:(876) 512-2680 x-mozilla-html:FALSE url:http://www.e2team.com version:2.1 end:vcard ___ 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] Rhino Channel Bank or ADIT 600
How do you integrate talk to the Asterisk server if you are using the cmg cards, and what is the cost difference with the CMG cards... C F wrote: On Tue, 22 Mar 2005 19:36:26 +, cmould [EMAIL PROTECTED] wrote: I am forklifting a Merridian option 51c with 112 Nortel Digital Handsets and 400 analog units. For the analog units I have quotes for 9 ADIT 600 48 port fxs units and 17 Rhino 24 port FXS channel banks. I have used neither. Which is the best choice? The price difference is not that great. I am looking at Citelinks 24 port Handset Gateway for the Nortel Digital units. (Any other suggestions would be appreciated). I'm actually trying to accomplish the same thing. 360 analog units, just hung up the phone with carrier access tech support, they where very helpful. plus this: http://lists.digium.com/pipermail/asterisk-users/2004-December/077099.html looks like I'm going with Adit. But instead of T1 from the Adit to * I plan on using CMG02 cards with the Adit 600, that gives me 9 Adit boxes, each one will have 5 FXS cards (5*8=40) and one CMG card, 9*40=360 FXS ports. That will make the Adit handle the bulk of the transcoding, and hence the CPU eat up. Also how many Asterisk servers would I need to handle 200 IP units in addition to the the above referenced legacy units? How do I size the server? Do I put voice mail on a different box? This is only a problem if you will be doing lots of transcoding (Zap -- SIP/G729 -- G711), if however you will be staying strictly VOIP and no codec transcoding (thats why I'm going with the CMG cards above, although it has to convert from MGCP to SIP, it doesn't eat up as much as from G711 to G729, or Zap to SIP), then you should't have a problem using one Dual Xeon box. If you must use telco provided T1s, you can either use another Adit 600 with a CMG on it, and hand it off to asterisk that way, or you could have one asterisk box just for the handling of the T1s, however asterisk with 4 T1s using a Digium quad T1 card, might (this is from experience, some people do have and others don't) have some echo problems. The other solution would be to have the 200 IP units connected to one box, and the analog ones connected to the other, and then use IAX from box to box, but I'm not sure it is better. I for myself am thinking of going with Quad Xeon boxes, an overkill? maybe. But I've never seen anybody crying for getting a better system than they need. Putting VM on a different box I don't think will accomplish anything, maybe make it even worse, since you will need the phone connected asterisk to bridge the call and open a stream to the voicemail box, maybe I'm wrong, but this is what I think. Also don't forget to look at this: http://www.voip-info.org/wiki-Asterisk+dimensioning Hope this helps, what ever your decision please put it on the list so others know about it. I plan on putting my installation on the wiki when it is done and running (another 3-4 months). Your comments much appreciated. ___ 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 begin:vcard fn:Carey Mould n:Mould;Carey org:E2 Systems Limited adr:237 Old hope Road;;Suite 11 12, technology Innovation Centre;Kingston;;Kgn 6;Jamaica email;internet:[EMAIL PROTECTED] title:CEO/Consultant tel;work:(876) 512-2680 x-mozilla-html:FALSE url:http://www.e2team.com version:2.1 end:vcard ___ 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] Rhino Channel Bank or ADIT 600
Thanks.. definately will keep the elist posted. Today I got a cost comparison from other PBX vendors and integrating the legacy phones. Nortel and Asterisk with the RHINO channel banks are similar (about US$ 36,000 in equipment costs). C F wrote: On Tue, 22 Mar 2005 19:36:26 +, cmould [EMAIL PROTECTED] wrote: I am forklifting a Merridian option 51c with 112 Nortel Digital Handsets and 400 analog units. For the analog units I have quotes for 9 ADIT 600 48 port fxs units and 17 Rhino 24 port FXS channel banks. I have used neither. Which is the best choice? The price difference is not that great. I am looking at Citelinks 24 port Handset Gateway for the Nortel Digital units. (Any other suggestions would be appreciated). I'm actually trying to accomplish the same thing. 360 analog units, just hung up the phone with carrier access tech support, they where very helpful. plus this: http://lists.digium.com/pipermail/asterisk-users/2004-December/077099.html looks like I'm going with Adit. But instead of T1 from the Adit to * I plan on using CMG02 cards with the Adit 600, that gives me 9 Adit boxes, each one will have 5 FXS cards (5*8=40) and one CMG card, 9*40=360 FXS ports. That will make the Adit handle the bulk of the transcoding, and hence the CPU eat up. Also how many Asterisk servers would I need to handle 200 IP units in addition to the the above referenced legacy units? How do I size the server? Do I put voice mail on a different box? This is only a problem if you will be doing lots of transcoding (Zap -- SIP/G729 -- G711), if however you will be staying strictly VOIP and no codec transcoding (thats why I'm going with the CMG cards above, although it has to convert from MGCP to SIP, it doesn't eat up as much as from G711 to G729, or Zap to SIP), then you should't have a problem using one Dual Xeon box. If you must use telco provided T1s, you can either use another Adit 600 with a CMG on it, and hand it off to asterisk that way, or you could have one asterisk box just for the handling of the T1s, however asterisk with 4 T1s using a Digium quad T1 card, might (this is from experience, some people do have and others don't) have some echo problems. The other solution would be to have the 200 IP units connected to one box, and the analog ones connected to the other, and then use IAX from box to box, but I'm not sure it is better. I for myself am thinking of going with Quad Xeon boxes, an overkill? maybe. But I've never seen anybody crying for getting a better system than they need. Putting VM on a different box I don't think will accomplish anything, maybe make it even worse, since you will need the phone connected asterisk to bridge the call and open a stream to the voicemail box, maybe I'm wrong, but this is what I think. Also don't forget to look at this: http://www.voip-info.org/wiki-Asterisk+dimensioning Hope this helps, what ever your decision please put it on the list so others know about it. I plan on putting my installation on the wiki when it is done and running (another 3-4 months). Your comments much appreciated. ___ 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 begin:vcard fn:Carey Mould n:Mould;Carey org:E2 Systems Limited adr:237 Old hope Road;;Suite 11 12, technology Innovation Centre;Kingston;;Kgn 6;Jamaica email;internet:[EMAIL PROTECTED] title:CEO/Consultant tel;work:(876) 512-2680 x-mozilla-html:FALSE url:http://www.e2team.com version:2.1 end:vcard ___ 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] Just saw your [Asterisk] xJack Segfault in Asterisk
Hi: Just saw your post while trying to solve a similar asterisk problem. Did not see any responses. Was your problem solved and what was the solution? Carey ___ 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] Cannot create mysql database with TRABAS
Hi all: I installed the TRABASS billing system and was successful in launchnig the app with my browser (mozilla). However I cannot create the mysql database from the configuration page. I have mysql running and can do so mannually. Can someone point me to a solution? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
