[asterisk-users] Aastra ignore call button hangs up call instead of going to voicemail
I have a Asterisk server (PBX in a Flash) with Aastra 57i phones. When
there is an incoming call the phone will display two buttons "answer"
and "ignore". If you press "ignore" the call is dropped instead of sent
to voice mail. The following is the log:
-- Called 111
-- SIP/111-1c14 is ringing
-- Got SIP response 486 "Busy Here" back from 192.168.3.126
-- SIP/111-1c14 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [...@macro-dial:8] Set("DAHDI/10-1", "DIALSTATUS=BUSY") in
new stack
-- Executing [...@macro-dial:9] GosubIf("DAHDI/10-1", "1?BUSY,1") in new
stack
== Spawn extension (macro-dial, s, 10) exited non-zero on 'DAHDI/10-1'
in macro 'dial'
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on
'DAHDI/10-1' in macro 'exten-vm'
== Spawn extension (from-did-direct, 111, 1) exited non-zero on
'DAHDI/10-1'
-- Hungup 'DAHDI/10-1'
The extensions.conf file has this macro-dial in it:
; Rings one or more extensions. Handles things like call forwarding and
DND
; We don't call dial directly for anything internal anymore.
; ARGS: $TIMER, $OPTIONS, $EXT1, $EXT2, $EXT3, ...
; Use a Macro call such as the following:
; Macro(dial,$DIAL_TIMER,$DIAL_OPTIONS,$EXT1,$EXT2,$EXT3,...)
[macro-dial]
exten => s,1,GotoIf($["${MOHCLASS}" = ""]?dial)
exten => s,n,SetMusicOnHold(${MOHCLASS})
exten => s,n(dial),AGI(dialparties.agi)
exten => s,n,NoOp(Returned from dialparties with no extensions to call
and DIALSTATUS: ${DIALSTATUS})
exten => s,n+2(normdial),Dial(${ds}) ;
dialparties will set the priority to 10 if $ds is not null
exten => s,n,Set(DIALSTATUS=${IF($["${DIALSTATUS_CW}"!
="" ]?${DIALSTATUS_CW}:${DIALSTATUS})})
exten => s,n,GosubIf($["${SCREEN}" != ""]?${DIALSTATUS},1)
exten => s,20(huntdial),NoOp(Returned from dialparties with hunt groups
to dial )
exten => s,n,Set(HuntLoop=0)
exten => s,n(a22),GotoIf($[${HuntMembers} >= 1]?a30) ; if this is from
rg-group, don't strip prefix
exten => s,n,NoOp(Returning there are no members left in the hunt group
to ring)
; dialparties.agi has setup the dialstring for each hunt member in a
variable labeled HuntMember0, HuntMember1 etc for each iteration
; and The total number in HuntMembers. So for each iteration, we will
update the CALLTRACE Data.
;
exten => s,n+2(a30),Set(HuntMember=HuntMember${HuntLoop})
exten => s,n,GotoIf($[$["${CALLTRACE_HUNT}" != "" ] &
$[$["${RingGroupMethod}" = "hunt" ] | $["${RingGroupMethod}" =
"firstavailable"] | $["${RingGroupMethod}" =
"firstnotonphone"]]]?a32:a35)
exten => s,n(a32),Set(CT_EXTEN=${CUT(FILTERED_DIAL,,$[${HuntLoop} +
1])})
exten => s,n,Set(DB(CALLTRACE/${CT_EXTEN})=${CALLTRACE_HUNT})
exten => s,n,Goto(s,a42)
;Set Call Trace for each hunt member we are going to call "Memory groups
have multiple members to set CALL TRACE For" hence the loop
;
exten => s,n(a35),GotoIf($[$["${CALLTRACE_HUNT}" != "" ] &
$["${RingGroupMethod}" = "memoryhunt" ]]?a36:a50)
exten => s,n(a36),Set(CTLoop=0)
exten => s,n(a37),GotoIf($[${CTLoop} > ${HuntLoop}]?a42) ; if this is
from rg-group, don't strip prefix
exten => s,n,Set(CT_EXTEN=${CUT(FILTERED_DIAL,,$[${CTLoop} + 1])})
exten => s,n,Set(DB(CALLTRACE/${CT_EXTEN})=${CALLTRACE_HUNT})
exten => s,n,Set(CTLoop=$[1 + ${CTLoop}])
exten => s,n,Goto(s,a37)
exten => s,n(a42),Dial(${${HuntMember}}${ds})
exten => s,n,Set(HuntLoop=$[1 + ${HuntLoop}])
exten => s,n,GotoIf($[$[$["foo${RingGroupMethod}" !=
"foofirstavailable"] & $["foo${RingGroupMethod}" !=
"foofirstnotonphone"]] | $["foo${DialStatus}" = "fooBUSY"]]?a46)
exten => s,n,Set(HuntMembers=0)
exten => s,n(a46),Set(HuntMembers=$[${HuntMembers} - 1])
exten => s,n,Goto(s,a22)
exten => s,n(a50),DBdel(CALLTRACE/${CT_EXTEN})
exten => s,n,Goto(s,a42)
; For call screening
exten => NOANSWER,1,Macro(vm,${SCREEN_EXTEN},BUSY,${IVR_RETVM})
exten => NOANSWER,n,GotoIf($["${IVR_RETVM}" != "RETURN" |
"${IVR_CONTEXT}" = ""]?bye)
exten => NOANSWER,n,Return
exten => NOANSWER,n(bye),Macro(hangupcall)
exten => TORTURE,1,Goto(app-blackhole,musiconhold,1)
exten => TORTURE,n,Macro(hangupcall)
exten => DONTCALL,1,Answer
exten => DONTCALL,n,Wait(1)
exten => DONTCALL,n,Zapateller()
exten => DONTCALL,n,Playback(ss-noservice)
exten => DONTCALL,n,Macro(hangupcall)
; make sure hungup calls go here so that proper cleanup occurs from call
confirmed calls and the like
;
exten => h,1,Macro(hangupcall)
Which unfortunately doesn't make much sense to me. I do see a
macro-exten-vm with a comment that it is where the call should be routed
is the extension is busy or doesn't answer. But I'm not sure how to
modify the macro-dial to make it happen.
I appreciate any help that anyone can give thanks in advance,
Jeremy
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[asterisk-users] Queue problem, ringing agents.
I'm running Asterisk 1.6.0.21 and Aastra 57i phones. I'm having an issue with the agent phones ringing when someone is in our queue. The first phone will ring 3 to 4 times then the call will roll over to the next phone as expected. However, any phone after the first one will only ring once and the wait between that phone and the next will be as if it if ringing 3 to 4 times. My queues.conf is as follows: [general] ; ; Global settings for call queues ; (none exist currently) ; ; Note that a timeout to fail out of a queue may be passed as part of application call ; from extensions.conf: ; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout]) ; example: Queue(dave|t|||45) #include queues_general_additional.conf #include queues_custom_general.conf [default] ; ; Default settings for queues (currently unused) ; #include queues_custom.conf #include queues_additional.conf #include queues_post_custom.conf queues_general_additional.conf: ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; persistentmembers=yes queues_custom_general.conf: is empty queues_custom.conf: is empty queues_additional.conf ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; [1001] announce-frequency=45 announce-holdtime=yes announce-position=yes autofill=yes eventmemberstatus=no eventwhencalled=no joinempty=yes leavewhenempty=no maxlen=0 music=default periodic-announce-frequency=0 queue-callswaiting=queue-callswaiting queue-thankyou=queue-thankyou queue-thereare=queue-thereare queue-youarenext=queue-youarenext retry=20 ringinuse=yes strategy=leastrecent timeout=20 weight=0 wrapuptime=0 queues_post_custom.conf: is empty Any help will be greatly appreciated, Jeremy -- _ -- 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] Detecting incoming faxes and forwarding to phone fax machine
I'm having a problem receiving incoming faxes and I'm hoping someone
here can help me out.
My system is a PBX in a Flash with one dahdi card for my incoming analog
lines and another dahdi card for my analog devices (fax and modem).
My dahdi-channels.conf file looks like:
; Autogenerated by /usr/sbin/dahdi_genconf on Tue Jun 23 14:56:24 2009
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is
intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global
settings
;
; Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER)
; Span 2: WCTDM/1 "Wildcard TDM800P Board 2"
;;; line="5 WCTDM/1/0 FXSKS"
signalling=fxs_ks
callerid=asreceived
faxdetect=incoming
group=0
context=from-zaptel
channel => 5
callerid=
group=
context=default
;AMPLABEL:Channel 5 - Button 1
;AMPWILDCARDLABEL(6):Trunks
;;; line="6 WCTDM/1/1 FXSKS"
signalling=fxs_ks
callerid=asreceived
faxdetect=incoming
group=0
context=from-zaptel
channel => 6
callerid=
group=
context=default
;;; line="7 WCTDM/1/2 FXSKS"
signalling=fxs_ks
callerid=asreceived
faxdetect=incoming
group=0
context=from-zaptel
channel => 7
callerid=
group=
context=default
;;; line="8 WCTDM/1/3 FXSKS"
signalling=fxs_ks
callerid=asreceived
faxdetect=incoming
group=0
context=from-zaptel
channel => 8
callerid=
group=
context=default
;;; line="9 WCTDM/1/4 FXSKS"
signalling=fxs_ks
callerid=asreceived
faxdetect=incoming
group=0
context=from-zaptel
channel => 9
callerid=
group=
context=default
;;; line="10 WCTDM/1/5 FXSKS"
signalling=fxs_ks
callerid=asreceived
faxdetect=incoming
group=0
context=from-zaptel
channel => 10
callerid=
group=
context=default
In FreePBX under Inbound Routes I have the following configured:
Fax Extension: 117
Fax Detection Type: Zaptel
Pause After Answer: 5
In FreePBX under General Settings I have the following configured:
Extension for fax machine for receiving faxes: 117
I can call into our IVR and and enter extension 117 and get the fax
machine. So I know the extension is working. However, when someone sends
me a fax I see the following in the logs:
[2010-01-19 10:29:25] VERBOSE[14688] app_macro.c: == Channel
'DAHDI/7-1' jumping out of macro 'from-zaptel-7'
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Executing
[7063277...@from-trunk:1] NoOp ("DAHDI/7-1", "Catch-All DID Match -
Found 7063277795 - You probably want a DID for this.") in new stack
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Executing
[7063277...@from-trunk:2]Goto("DAHDI/7-1", "ext-did,s,1") in new stack
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Goto (ext-did,s,1)
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Executing
[...@ext-did:1] Set("DAHDI/7-1", "__FROM_DID=s") in new stack
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Executing
[...@ext-did:2] ExecIf("DAHDI/7-1", "1 ?Set(CALLERID(name)=8009806858)) in
new stack
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Executing
[...@ext-did:3] Set("DAHDI/7-1", "FAX_RX=117") in new stack
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Executing
[...@ext-did:4] Set("DAHDI/7-1", "fax_rx_email=jwin...@logicalsi.com") in
new stack
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Executing
[...@ext-did:5] Answer("DAHDI/7-1", "") in new stack
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Executing
[...@ext-did:6] Wait("DAHDI/7-1", "5") in new stack
[2010-01-19 10:29:26] VERBOSE[14688] chan_dahdi.c: -- Redirecting
DAHDI/7-1 to fax extension
[2010-01-19 10:29:26] VERBOSE[14688] pbx.c: == Spawn extension
(ext-did, fax, 1) exited non-zero on 'DAHDI/7-1'
[2010-01-19 10:29:26] VERBOSE[14688] pbx.c: -- Executing
[...@ext-did:1] Goto("DAHDI/7-1", "ext-fax,in_fax,1") in new stack
[2010-01-19 10:29:26] VERBOSE[14688] pbx.c: -- Goto
(ext-fax,in_fax,1)
[2010-01-19 10:29:26] VERBOSE[14688] pbx.c: -- Executing
[in_...@ext-fax:1] StopPlayTones("DAHDI/7-1", "") in new stack
[2010-01-19 10:29:26] VERBOSE[14688] pbx.c: -- Executing
[in_...@ext-fax:2] GotoIf("DAHDI/7-1", "0?3:analog_fax,1") in new stack
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Executing
[...@ext-did:2] ExecIf("DAHDI/7-1", "1 ?Set(CALLERID(name)=8009806858)")
in new stack
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Executing
[...@ext-did:3] Set("DAHDI/7-1", "FAX_RX=117") in new stack
[2010-01-19 10:29:25] VERBOSE[14688] pbx.c: -- Executing
[...@ext-did:4] Set("DAHDI/7-1", "FAX_RX_ZAP/1,20,d") in new stack ZAP'
e 'ZAP' (cause 66 - Channel not implemented)
at this time (1:0/0/1)
") in new stack
g_fax, 4) exited non-zero on 'DAHDI/7-1'
e application/pdf --file ") in new stack
") in new stack
exited non-zero on 'DAHDI/7-1'
[2010-01-19 10:29:27] VERBOSE[14688] chan_dahdi.c: -- Hungup
'DAHDI/7-1'
To my untrained eye, I would say this stems from the ZAPTel becoming
DAHDI problem. But I'm clueless how to fix it.
Any help will
Re: [asterisk-users] q: which Browser-GUI do u guys use?
While I agree with Steve on a philosophical level, there are a lot of merits to command lines and direct editing of configuration, there also comes a time when "just getting the job done" is benefited by a nice point-n-click. I have found in my career that I may spend a month neck deep in a project, such as implementing Asterisk, then for the following 6 months never have to touch it again. During those 6 months away, I would have been implementing a new intrusion prevention system, probably doing a bit of programming, managing my 300+ Linux servers, or helping our DBA setup new MS-SQL clusters. When I'm asked to do something like, say reroute all incoming calls through a new IVR with several new queues, it sure helps to have a gui to help out instead of having to relearn the guts of the system. But these are just my thoughts on the subject. And so far during my month of being neck deep in implementing Asterisk I have used FreePBX. Jeremy > None. I'm a command line weenie. > > ) GUIs don't let you annotate your changes -- who did what (or what they > thought they were doing), when, and why. > > ) GUIs don't support any sort of "versioning." > > ) GUIs don't support any sort of configuration rollback. > > All of these are essential when something that used to work suddenly > doesn't. (Sometimes, client's don't notice something isn't working for > months -- way beyond my short term memory.) > > I'm sure I could come up with dozens more, these were just the first 3. > (Probably not even the most important 3.) > > Oh. Here's 1 more -- GUIs impede truly understanding a system. > ___ -- 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] Resetting Day/Night setting
All of this is setup using FreePBX and to be honest, with all of the macros that FreePBX adds to the *.conf files I'm not really sure what Asterisk is doing. I've only been playing with all of this for about two weeks now and most of that was waisted trying to figure out the ZAPTEL vs DAHDI stuff. My planned incoming routes will look something like this: (Incoming) | /\ /\ / \ / \ / if \_After_/Day\\__Day \time/ Hours \Nite/ | \ / \ / | \/ \/ | |Work|Night | /\ Hours \/ \/ / \ - - /Day\\_Day__>(Queue)(VoiceMail) \Nite/- - \ / /\ \/| |Night | -- I left out all of the announcements and IVR because I "love" drawing in ASCII so much, but you get the general idea. With the Time Condition (if time) we will have automated routing of the calls to the queue during work hours and voicemail after hours. However, there are a few times a year when we need to overrule this time condition for situations where tech support is offered on a weekend or we are closed on a holiday. My thought was instead of constantly having to update the Time Condition, we could use the Day/Night Control which gives a '*280' dial code. The problem is in the case of say a holiday on a Friday, when the tech support manager decides at the last minute to let his people go early and dials *280 and then leaves. Come Monday the override will still be in place and since call volumes are usually low in the morning, it could be noon or later before someone realizes something is wrong. My thought was to use cron to run a script that will check the status of the Day/Night Control and compare it with the Time Condition and if they match, set the Day/Night Control back to default (day). So in our holiday scenario, come 5:00pm that Friday night when the Time Condition would switch to "after hours" the Day/Night Control switches back to its default setting. Being new to voice systems, I knew how I would handle it as a Linux Sysadmin, I was curious how "telecom guys" would go about it. To answer your question, I believe FreePBX is using a GotoIfTime clause for the Time Condition but I'm not exactly sure. I'm more worried about giving our tech support manager the ability to override the "normal" dial plan without having to call me. Thanks again, Jeremy On Tue, 2009-07-07 at 11:32 -0400, Jared Smith wrote: > On Tue, 2009-07-07 at 10:47 -0400, Jeremy Winder wrote: > > It seemed to me cron was going to be the best solution. > > Sounds like overkill to me... why not just use a GotoIfTime clause in > your dialplan? > > ___ -- 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] Resetting Day/Night setting
It seemed to me cron was going to be the best solution. I'll create a script that will run every minute or so that compares the Day/Night Control with the Time Condition and when their outcomes match, reset the Day/Night Control back to default. I was curious as to how others would approach the same problem. Thanks again, Jeremy On Tue, 2009-07-07 at 09:14 -0500, Danny Nicholas wrote: > I concur that this is probably the best solution; We could provide more > helpful answers with more question details. There are usually at least two > "correct" solutions to any query you can post; the more detail you can > provide, the more likely you are to get a correct and efficient answer. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Godbout > Sent: Tuesday, July 07, 2009 9:08 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Resetting Day/Night setting > > Use a cron entry to reset the setting using a cli command, as long as the > setting is in the internal database of asterisk. > > > -Original Message- > > From: jwin...@logicalsi.com > > Sent: Tue, 07 Jul 2009 09:44:58 -0400 > > To: asterisk-users@lists.digium.com > > Subject: [asterisk-users] Resetting Day/Night setting > > > > I'm not sure if this is part of Asterisk or FreePBX so I apologize if > > this is the wrong list to ask my question. > > > > As part of my companies call routes, I have a Time Condition for our > > tech support queue. I would like to add a Day/Night Control so the Time > > Condition can be overruled. However, I'm afraid someone will forget to > > turn it off again. What is the best way of resetting this control back > > to its default setting at say midnight? > > > > Thanks in advance, > > > > Jeremy > > > > > > ___ > > -- 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 > > > Receive Notifications of Incoming Messages > Easily monitor multiple email accounts & access them with a click. > Visit http://www.inbox.com/notifier and check it out! > > ___ > -- 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
[asterisk-users] Resetting Day/Night setting
I'm not sure if this is part of Asterisk or FreePBX so I apologize if this is the wrong list to ask my question. As part of my companies call routes, I have a Time Condition for our tech support queue. I would like to add a Day/Night Control so the Time Condition can be overruled. However, I'm afraid someone will forget to turn it off again. What is the best way of resetting this control back to its default setting at say midnight? Thanks in advance, Jeremy ___ -- 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] Extension status as XML for an Aastra 57i
So what I'm gathering is this I have to map each extension to a button, whether physical on a 560m or 536m or virtual using the soft buttons on the phone. What I was hoping for was something like the Directory app http://voip-pbx/aastra/directory.php that came with the phone's firmware that shows everyone's extensions, just expanded to show that extension's status. Has anyone done this, is it even possible? Thanks again, Jeremy On Wed, 2009-07-01 at 21:11 -0500, Jonathan Moore wrote: > On Wed, Jul 1, 2009 at 4:40 PM, Olivier wrote: > > True but how can a single light be blinking because extension 1001 is > > receiving a call and at the same time, be turned on because extension 1002 > > is on call ? > > Maybe typing on Next button would alternatively show extension 1001 or 1002 > > status, but without a press on this Next key, a user can't be aware of all > > status changes as he would if equiped with dedicated BLF. > > That's the problem with using the buttons in this manner. Exactly as > you said, the > light will only represent a single item at any one time. To see more than one > you have to switch what the light is representing at that moment. > > Of course, as you point out this may not work out. If that's the > case, your next > option is to go with the 560m or 536m depending. Of course, you can also add > several of to each 57i phone (up to 3, IIRC). > > > But of course, this might be enough for some (operators, ...) but not for > > all (group secretary ...). > > No disagreement. It all comes down to how much you're willing to pay for > the convenience. If you (or the users) don't want to switch screens, you get > a single, or more than one, sidecar. > > -jonathan > > ___ > -- 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
[asterisk-users] Extension status as XML for an Aastra 57i
I'm in the process of converting our current hybrid key system to Asterisk and Aastra 57i phones. One of the features that seems to be a show stopper for almost everyone in the office is the inability to see who is on the phone. Can someone point in the right direction to setup an XML app on the phone so they can press a soft-button and get a list of extensions and their statuses? I know I can use BLF and the line 2-4 buttons; but there are a lot more then 3 other people working here and I'm planning on using those of parking lots. Any help will be greatly appreciated as I'm an Asterisk noob learning as fast as I can. Thanks in advance, Jeremy ___ -- 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] Different inbound routes for each interface on a TDM800P card.
I'm new to Asterisk and inherited this project so I apologize if this question has been asked a hundred time before. I did start with Google but I may not be asking the right questions, because I wasn't finding any answers. I have Asterisk 1.4.24 and FreePBX 2.5 running and using a Digium TDM800P to interface with our six analog phone lines from the telco. Currently I have a single trunk setup ZAP/G0 that will allow all incoming call go to your IVR and allow outgoing calls. My problem is I need two of those lines to be routed to one IVR for our tech support people and the other four to be routed to our main business IVR. For the life of me, I can't seem to find anything online about how to do this. I thought about setting up six separate trunks; but I didn't see anywhere you specify the channel for the trunk and right now my single trunk is working with all of the analog lines. Any help will be greatly appreciated, Jeremy ___ -- 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
