Re: [asterisk-users] Route an incoming call by ANI*DNIS
Thank you all,
It just so turns out that it was a bad zaptel module. We saw
another post on digiums site where someone was having the exact same
problem with several versions of zaptel. We changed to the one that he
said worked (1.2.21), and all is well now. (And asterisk is now parsing
the ani and dnis properly).
Tilghman Lesher wrote:
On Tuesday 20 November 2007 02:38:38 Atis Lezdins wrote:
I won't be able to help with hardware part, but there's a simple trick
to get them as you want:
[incoming]
_X.,1,Set(DNIS=${CUT(${EXTEN:-4})})
_X.,2,Goto,dnis,${DNIS},1
[dnis]
6789 = ...
I don't think you've actually tested this, because if you had, you would find
that it does not work.
[incoming]
exten = _X.,1,Goto(dnis,${EXTEN:-4},1)
[dnis]
exten = 6789,1,.
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Re: [asterisk-users] Route an incoming call by ANI*DNIS
Sorry for my very delayed response. To answer a few questions: 1. Right, the *ANI*DNIS* is not working correctly. When the telco sends it, we are always missing the beginning of it. I almost always get a 7 digit ani, but sometimes it is 8 or 6. 2. The phone company assured me several times that we are not set up for feature group d. I tried anyway and it had the same affect. Got a non-Feature Group D input on channel 77. Assuming EM Wink instead.. Pretty smart. 3. I've also triad featb featdmf and featdmf_ta. All of which were the same or much works. Featb atcually crashed my polycom phone when I tried to dial into the pbx. :) 4. I didn't try 5ess as were are not using a pri. After this whole ordeal, where contemplating having all our t1's switched to pri. That would really be a bad thing, but its a pretty drastic measure. 5. The callerid issue is partially solved. I handled it with CALLERID=$DID. The main phone system runs a perl script to parse, and handle the rest of it. Which actually I like this better, because I don't have to setup extensions for every DNIS.. I have about 50 of them. We are still having the same signaling issue with both XO and GBX. Heres the problem I'm getting in asterisk. 1. A wink issue Nov 19 05:35:06 DEBUG[12565] chan_zap.c: Got wink in weird state 4 on channel 92 2. Calls that drop randomly Nov 19 12:29:51 DEBUG[15370] channel.c: Avoiding initial deadlock for 'SIP/1026-b7901568' Nov 19 12:31:02 DEBUG[22546] channel.c: Didn't get a frame from channel: SIP/1026-b7901568 Nov 19 12:31:02 DEBUG[22546] channel.c: Bridge stops bridging channels SIP/1026-b7901568 and Zap/1-1 3. Two phone companies that say I'm sending too many winks went the seizure, and a Digium rep who says it should just work. I am shot... Surely I can't be the only person who has this problem. Is there anyone else not using PRI, who had issues like this.?? Dan Jon Weisman wrote: Dan, What happen w/ this? Did you figure it out? I've setup w/ XO using ESF/B8ZS, and 5ESS for the switchtype, worked great and got the ANI as well. I dont think you can get ANI on EM Wink trunks, how about feature group d? -Jon - Original Message - From: Dan Casey [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, November 02, 2007 9:47 AM Subject: [asterisk-users] Route an incoming call by ANI*DNIS does anyone know how to route a call coming in with ANI*DNIS* Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing Set(Zap/49-1, DID=1231234*4812*) in new stack I tried making a route for _.*4812* but that matched everything rather then just the dnis i wanted.. any ideas? I would preferably like pass the callerid along to my extensions, but for now the important thing is routing. Thanks ___ --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] Route an incoming call by ANI*DNIS
does anyone know how to route a call coming in with ANI*DNIS* Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing Set(Zap/49-1, DID=1231234*4812*) in new stack I tried making a route for _.*4812* but that matched everything rather then just the dnis i wanted.. any ideas? I would preferably like pass the callerid along to my extensions, but for now the important thing is routing. Thanks ___ --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] Route an incoming call by ANI*DNIS
for this t1 my settings are d4,ami sf em_w I was starting to suspect that em_w didn't support ANI.. I was hoping that this wasn't the case as the telco told me to use em wink. I just went through wink nightmare with XO and i'm dreading repeating it.. I'll give it a try, thanks! BJ Weschke wrote: Dan Casey wrote: does anyone know how to route a call coming in with ANI*DNIS* Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing Set(Zap/49-1, DID=1231234*4812*) in new stack I tried making a route for _.*4812* but that matched everything rather then just the dnis i wanted.. any ideas? I would preferably like pass the callerid along to my extensions, but for now the important thing is routing. This shouldn't be necessary. I think with the correct signaling set in zaptel.conf, chan_zap should be doing the parsing work for you. What signaling are you set for now? You might want Feature Group D vs. EM_Wink? ___ --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] Proper trunk to connect two systems.
Hello, I am replacing an exisiting call center with a new asterisk based solution. This will initially consist of to phone servers. The first being the main PBX, and the second being a predictive dialer. The dialer will have sip extensions for all the agents, while the main pbx will hand pretty much everything else. The two boxes will be right next two each other, and are currently connected via an IAX2 trunk. All manually made phone calls work with no problem. There is an issue however with the dialer software (vicidial) using an IAX trunk. It is a little finicky sometimes leaves iax in a state where it cannot resolve it's channel name and drops the call. I haven't spent any time really troubleshooting this yet, but apparently it does work after poking at the settings for a while. Before I bother troubleshooting IAX, I figured I would ask some of the more knowledgeable folks here about what is the best way to connect the two servers. My options as far as I know are: 1. Play with IAX2 until it works. 2. Create SIP trunks instead. 3. TDMoE and treat it as zap. (I should mention that only the main pbx has digium hardware. The dialer uses ztdummy). This connection between the two servers will need to support a minimum of 35 concurrent calls, to eventually 200 concurrent calls. At that point of course I'll probably be looking at biocluster or other redundant setups. I am currently leaning towards TDMoE. If I'm figuring this correctly a gigabit crossover connection would give me the equivalent of 500 E1 circuits? I wouldn't push it that far, but what would be a reasonable amount to push on it. Any thoughts? -dc ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Force SIP hang up.
Is there a way to hang up on a sip channel. One of my phones is saying it's busy while it's not (even after rebooting it). I logged into asterisk, and did a sip show channel 232, and sure enough it thinks it's on a call. How can I force it to close? ___ --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] SIP asterisk over Linksys VPN
I tried to do sip over vpn with with a linksys router handle just one phone. When I tried it, it worked fine. Once i shipped it out we had all types of problems. at first it was fine, then 1 out of 5 calls would sound like cell phones. Now I can call him be he can't hear anything. Everything else works fine through the vpn. Most importantly, I can't trouble shoot correctly. I finally gave up and got him callvantage. Now all I have to worry about is forwarding a DID number. Raymond McKay wrote: I usually run the RV series of router for this. Much better thoughourput on the VPN. Remember these low end devices can usually only handle about 1Mbps - 3Mbps of encryption max depending on the unit. Other than that, I have had up to 8 behind a VPN such as this. I do generally recommend though that a small appliance style asterisk box sit on any side of a remote connection with a 1 port FXO card installed for timing, emergency 911 capability, and trucking and jitterbuffer support over IAX2. This, IMHO, tends to provide for better reliability. I generally recommend some kind of HDD less Compact Flash based system. Less mechanicals to break and you can pick one up generally for $600-$800 with the digium card depending on speed and number of phones to support. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 15, 2006 1:23 PM Subject: [asterisk-users] SIP asterisk over Linksys VPN Has anybody tried using a VPN and around 10 phones behind the tunnel to connect to an asterisk server using Linksys VPN routers? Like this one: http://www.linksys.com/servlet/Satellite?c=L_Product_C2childpagename=US%2FLayoutcid=1115416832495pagename=Linksys%2FCommon%2FVisitorWrapper ___ --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
[asterisk-users] macro-dialout without specifying trunk
I am trying to have a SIP extension that will dial an outside phone
number (ie: cell phone) using a zap channel.
I am using the following hack, which doesn't technically works, but not
nicely. What i want to do is have it pick an available trunk from zap1
to zap20.
I have tried using dialout,s,number and also dialout,g1,number i
just keep getting all circuits busy. (I have posted my zapata.conf below).
I can do it if I specify the specific trunk. Here is my extension:
exten = 299,1,Macro(dialout-return,1,1914426)
exten = 299,2,Macro(dialout-return,2,1914426)
exten = 299,3,Macro(dialout-return,5,1914304)
exten = 299,4,hangup
; dialout-return. Like dialout but doesn't go to outisbusy.
[macro-dialout-return]
exten = s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4);check
for CID override for exten
exten = s,2,SetCallerID(${ECID${CALLERIDNUM}})
exten = s,3,Goto(6)
exten = s,4,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6);check
for CID override for trunk
exten = s,5,SetCallerID(${OUTCID_${ARG1}})
exten = s,6,SetVar(length=${LEN(${DIAL_OUT_${ARG1}})})
exten = s,7,Dial(${OUT_${ARG1}}/${ARG2:${length}})
zapata.conf
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=pri_cpe
switchtype=national
rxwink=300 ; Atlas seems to use long (250ms) winks
callerid=asreceived
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming
channel = 1-7
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Re: [asterisk-users] Re: Re: How to forward a call to an outside line
Funny you mention it. The problem that i was having with the trunks was
that i put every channel in it's own group.
In amp, all the trunks where identified by g0,g1,g2,g3. We changed
that to 0,1,2,3 yesterday. Doing that it allows me to dial out on any
one trunk that i want, where before i could only dial on the first one.
Looking in zapata.conf i have
[trunkgroups]
nothing here
[channels]
group=1
callgroup=1
pickupgroup=1
So i think i see where i need to go with that. This is the problem
with using AAH vs. setting it up manually. I didn't learn how it
really works.
Thanks a lot!
Steven wrote:
If you have zap groups defines then
instead of
exten = 299,1,Macro(dialout,5,1914426,,)
you would use
exten = 299,1,Macro(dialout,g0,1914426,,)
g0 or gwhatever is the group number
for a list of zap trunks.
Look at group in zapata.conf.
--
--
Steven
http://www.glimasoutheast.org
"Dan Casey" [EMAIL PROTECTED]
wrote in message news:[EMAIL PROTECTED]...
I actually have it semi-working. My trunks were set up improperly.
Now i can do it, but only if i specify a specific zap channel.
exten = 299,1,Macro(dialout,2,1914304,,)
the 2 takes me to zap 1. I tried to replace that with "s" but no luck..
any idea how to do this where it will pick any available trunk?
I gave that a try but had no luck. I keep getting all circuits busy.
Perhaps there is another way.
I think it is having trouble when transfering zap to zap.
but no matter what i do i can't get it.
I made a sip number to try from, but its not working
[ext-local-custom]
;exten = 299,1,Macro(dialout-trunk,0,914426,,)
exten = 299,1,Macro(dialout,5,1914426,,)
This is what i've been getting all day long.
-- Executing Dial("SIP/212-ace5", "ZAP/g4/1914426") in new stack
== Everyone is busy/congested at this time
I assure you are not busy :)
Steven wrote:
I use :
exten = 5600,3,Macro(dialout-trunk,0,91248XXX,,)
Mind you that I am using FreePBX, so I am using the dialout-trunk macro.
But before I used FreePBX, I would do the same with Dial.
5600 is a DID number that gets forwarded outside.
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Re: [asterisk-users] Re: How to forward a call to an outside line
I gave that a try but had no luck. I keep getting all circuits busy. Perhaps there is another way. I think it is having trouble when transfering zap to zap. but no matter what i do i can't get it. I made a sip number to try from, but its not working [ext-local-custom] ;exten = 299,1,Macro(dialout-trunk,0,914426,,) exten = 299,1,Macro(dialout,5,1914426,,) This is what i've been getting all day long. -- Executing Dial(SIP/212-ace5, ZAP/g4/1914426) in new stack == Everyone is busy/congested at this time I assure you are not busy :) Steven wrote: I use : exten = 5600,3,Macro(dialout-trunk,0,91248XXX,,) Mind you that I am using FreePBX, so I am using the dialout-trunk macro. But before I used FreePBX, I would do the same with Dial. 5600 is a DID number that gets forwarded outside. ___ --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] Re: How to forward a call to an outside line
I actually have it semi-working. My trunks were set up improperly.
Now i can do it, but only if i specify a specific zap channel.
exten = 299,1,Macro(dialout,2,1914304,,)
the 2 takes me to zap 1. I tried to replace that with "s" but no luck..
any idea how to do this where it will pick any available trunk?
I gave that a try but had no luck. I keep getting all circuits busy.
Perhaps there is another way.
I think it is having trouble when transfering zap to zap.
but no matter what i do i can't get it.
I made a sip number to try from, but its not working
[ext-local-custom]
;exten = 299,1,Macro(dialout-trunk,0,914426,,)
exten = 299,1,Macro(dialout,5,1914426,,)
This is what i've been getting all day long.
-- Executing Dial("SIP/212-ace5", "ZAP/g4/1914426") in new stack
== Everyone is busy/congested at this time
I assure you are not busy :)
Steven wrote:
I use :
exten = 5600,3,Macro(dialout-trunk,0,91248XXX,,)
Mind you that I am using FreePBX, so I am using the dialout-trunk macro.
But before I used FreePBX, I would do the same with Dial.
5600 is a DID number that gets forwarded outside.
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Re: [asterisk-users] Re: How to forward a call to an outside line
Greg Broiles wrote:
On 8/3/06, Dan Casey [EMAIL PROTECTED] wrote:
I actually have it semi-working. My trunks were set up improperly.
Now i can do it, but only if i specify a specific zap channel.
exten = 299,1,Macro(dialout,2,1914304,,)
the 2 takes me to zap 1. I tried to replace that with s but no
luck.. any
idea how to do this where it will pick any available trunk?
Perhaps someone who uses Zap frequently has a better idea, but I would
be inclined to approach this as a case where you'd want to try
multiple devices in serial. This is an (admittedly ugly) example of my
tollfree dialing macro, which tries several providers in the event
that one is unable to complete the call for whatever reason.
[macro-tollfree]
exten = s,1,dial(${TRXTEL}/${ARG1}-noads,90,tr)
exten = s,2,dial(${NUFONE}/${ARG1},90,tr)
exten = s,3,dial(${TELIAX}/${ARG1},90,tr)
exten = s,4,dial(${TRUNK3}/${ARG1},90,tr)
exten = s,5,Congestion
Also - in case it's useful - this is a bit of my dialplan that I used
to ring a local extension, then call an outside answering service if
the local extension doesn't answer -
exten = 800936,1,Dial(${RON},20,r)
exten = 800936,2,Dial(${TRUNK}/${ANSWERINGSERVICE},60,r)
.. my application used IAX, not Zap, for the outgoing call, but it
seems like the behavior should be similar.
I was thinking of doing that, but was trying to avoid it. Right now I'm
only using 7 channels on our T1 for voice, so it's not too bad. I'm
assuming it should be under a 2 second delay even if it has to try every
single channel..
Thanks a lot for your help, this was driving me nuts today.
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[Asterisk-Users] Can't hear auto-attendant
Using [EMAIL PROTECTED] 1.1 and 1.3 ip phones are on the same network as asterisk. I can call another extension and talk/listen w/ no problem. If i dial my own extension (or do anything that makes asterisk play sounds back to me ) I cannot hear anything. this is what shows up in the CLI, when i dial my own extension. -- Goto (macro-vm,s-BUSY,1) -- Executing VoiceMail(SIP/440-35ab, [EMAIL PROTECTED]) in new stack -- Playing 'vm-theperson' (language 'en') == Spawn extension (macro-vm, s-BUSY, 1) exited non-zero on 'SIP/440-35ab' in macro 'vm' == Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/440-35ab' in macro 'exten-vm' == Spawn extension (from-internal, 440, 1) exited non-zero on 'SIP/440-35ab' -- Executing Macro(SIP/440-35ab, hangupcall) in new stack Thanks, Dan ___ 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
