I have installed a DIGIUM 1TDM410PLF on a ST V1.6.
I have set Channel 1 up as an FXS and it is working properly.
I have 2-3 set as FXO. I set up a channel to DID map on Channel 2 and then set up an inbound route to an SIP extension.
When I call the pstn number connected to the FXO port I get the following output.
2011-04-27 15:18:39] NOTICE[20116]: chan_dahdi.c:8786 ss_thread: Got event 18 (Ring Begin)...
[2011-04-27 15:18:40] NOTICE[20116]: chan_dahdi.c:8786 ss_thread: Got event 2 (Ring/Answered)...
[2011-04-27 15:18:44] NOTICE[20116]: chan_dahdi.c:8786 ss_thread: Got event 18 (Ring Begin)...
-- Executing [s@from-outside:1] Wait("DAHDI/2-1", "1") in new stack
-- Executing [s@from-outside:2] Set("DAHDI/2-1", "__INCOMINGCLI=") in new stack
-- Executing [s@from-outside:3] Goto("DAHDI/2-1", "from-outside-redir,s,1") in new stack
-- Goto (from-outside-redir,s,1)
-- Executing [s@from-outside-redir:1] Set("DAHDI/2-1", "DIALED_NUMBER=s") in new stack
-- Executing [s@from-outside-redir:2] GotoIf("DAHDI/2-1", "0?s,13") in new stack
-- Executing [s@from-outside-redir:3] GotoIfTime("DAHDI/2-1", "17:00-23:59,*,*,*?from-outside-s-tl-off-hours,s,1") in new stack
-- Executing [s@from-outside-redir:4] GotoIfTime("DAHDI/2-1", "0:00-8:59,*,*,*?from-outside-s-tl-off-hours,s,1") in new stack
-- Executing [s@from-outside-redir:5] GotoIfTime("DAHDI/2-1", "*,sat-sun,*,*?from-outside-s-tl-off-hours,s,1") in new stack
-- Executing [s@from-outside-redir:6] GotoIfTime("DAHDI/2-1", "*,*,1,jan?from-outside-s-tl-off-hours,s,1") in new stack
-- Executing [s@from-outside-redir:7] GotoIfTime("DAHDI/2-1", "*,*,31,may?from-outside-s-tl-off-hours,s,1") in new stack
-- Executing [s@from-outside-redir:8] GotoIfTime("DAHDI/2-1", "*,*,4,jul?from-outside-s-tl-off-hours,s,1") in new stack
-- Executing [s@from-outside-redir:9] GotoIfTime("DAHDI/2-1", "*,*,6,sep?from-outside-s-tl-off-hours,s,1") in new stack
-- Executing [s@from-outside-redir:10] GotoIfTime("DAHDI/2-1", "*,*,24-25,nov?from-outside-s-tl-off-hours,s,1") in new stack
-- Executing [s@from-outside-redir:11] GotoIfTime("DAHDI/2-1", "*,*,24-25,dec?from-outside-s-tl-off-hours,s,1") in new stack
-- Executing [s@from-outside-redir:12] GotoIfTime("DAHDI/2-1", "*,*,31,dec?from-outside-s-tl-off-hours,s,1") in new stack
-- Executing [s@from-outside-redir:13] NoOp("DAHDI/2-1", "s") in new stack
-- Executing [s@from-outside-redir:14] GotoIfTime("DAHDI/2-1", "*,*,*,*?from-outside-s-tl-allhours,s,1") in new stack
-- Goto (from-outside-s-tl-allhours,s,1)
-- Executing [s@from-outside-s-tl-allhours:1] Macro("DAHDI/2-1", "tl-menu,tl-main-menu-open") in new stack
-- Executing [s@macro-tl-menu:1] Set("DAHDI/2-1", "CALLERID(name)=") in new stack
-- Executing [s@macro-tl-menu:2] Goto("DAHDI/2-1", "tl-main-menu-open,s,1") in new stack
-- Goto (tl-main-menu-open,s,1)
== Channel 'DAHDI/2-1' jumping out of macro 'tl-menu'
-- Executing [s@tl-main-menu-open:1] GotoIf("DAHDI/2-1", "0?start") in new stack
-- Executing [s@tl-main-menu-open:2] Set("DAHDI/2-1", "TL_LEVEL=1") in new stack
-- Executing [s@tl-main-menu-open:3] Ringing("DAHDI/2-1", "") in new stack
-- Executing [s@tl-main-menu-open:4] Wait("DAHDI/2-1", "5") in new stack
-- Executing [s@tl-main-menu-open:5] Answer("DAHDI/2-1", "") in new stack
[2011-04-27 15:18:51] WARNING[20116]: chan_dahdi.c:2685 dahdi_enable_ec: Unable to enable echo cancellation on channel 2 (No such device)
-- Executing [s@tl-main-menu-open:6] Set("DAHDI/2-1", "TIMEOUT(digit)=10") in new stack
-- Digit timeout set to 10.000
-- Executing [s@tl-main-menu-open:7] Set("DAHDI/2-1", "TIMEOUT(response)=15") in new stack
-- Response timeout set to 15.000
-- Executing [s@tl-main-menu-open:8] BackGround("DAHDI/2-1", "ogm/sampleopen") in new stack
-- Playing 'ogm/sampleopen.slin' (language 'en')
[2011-04-27 15:18:51] WARNING[20116]: file.c:1286 waitstream_core: Unexpected control subclass '2'
[2011-04-27 15:18:51] WARNING[20116]: file.c:1286 waitstream_core: Unexpected control subclass '2'
[2011-04-27 15:19:02] WARNING[20116]: pbx.c:4442 __ast_pbx_run: Don't know what to do with 'DAHDI/2-1'
-- Hungup 'DAHDI/2-1'
-- Remote UNIX connection disconnected
and the call does not ring where I want it to go?
any thoughts.
s is not the destination but
s is not the destination but the last resort. we are trying to map a dahdi channel to a did since we have no dialed number on these inbound fxo line
if you want to argue then you
if you want to argue then you can figure it out yourself. I tried to help you but apparently you're too arrogant to even listen. Some advise, when you're about to tell a grand master that they're wrong you better have your facts straight.
Theres no DID outpulsed by the telephone company on analog lines, so you have to use 's' in your dialplan. 's' is not 'last resort', its for when the extension is not declared. In order to use the channel-to-did map you must do it via the 's' extension.
Very Very Sorry
My comment was not suppose to come across as arrogant. It was suppose to be a question. I didn't take my time to word it properly. This is my first ever experience with Astrisk and I have only been working on it for one week now. So I would like to say thank you for taking the time to respond, I would also like to say sorry for coming across as arrogant. I am not at all. I admit I do no know anything at the present time other than what I have taught myself.
so how I should word this is. I don't know where to set the script for the s extension. If your still willing to help I would appreciate it. I see your name all over the forums and I know you know your stuff. I hope I can learn from you in the future if your willing to help.
Once again Sorry!
Chris
create an s extension if you
create an s extension if you do not already still have one defined.. keep the all-hours schedule and change the script to 'tl-reroute-analog'
that will execute a Goto statement based on how you created your channel-to-did map.
Thank you very much. This
Thank you very much. This makes sense now. What happened was I was using the evaluation version of Thirdline ST. It was acting very strange. I would add routes and trunks then delete and it was acting very weird. I purchased the licence today and now it is going a lot better.
I will give it a shot and let you know!
Thank you and once again sorry for the way I came across.
more details please
What does this mean?
can I assign a certain did that points to the s extension?
I am running the MultiTennant version
and how would I do it
Thank you for any help
analog line
I have been using the tdm400p card witht he mte edition for awhile
I have just installed the newest version of mte from your web site and I got it to mostly work but when I try to forward a call for multiple extensions the call just hangs up.
I was hoping to make sure my hack was intergrated better.
If I cant use the analog card is there a better way that does not cost a lot but with good uptime?
I also use some teliax form a voip provider but I am having an issue with them right now with calls coming in
I keep getting this error
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
[2011-07-24 15:05:20] NOTICE[4568]: chan_sip.c:21515 handle_request_invite: Failed to authenticate device "WIRELESS CALLER" ;tag=KXy78a5gay8KK
[2011-07-24 15:05:27] WARNING[4568]: chan_sip.c:3551 retrans_pkt: Retransmission timeout reached on transmission d66daf11-30e3-122f-32bd-00114336bd5a for seqno 15436491 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
I like using the analog lines incase the internet goes down since they have the best uptime.
I would be happy to pay for a good analog intergration - The tdm400p auto loads with thirdlane
Not sure what to do kind of in a pickle
you didnt pick the right script for the s extension