[Asterisk-Users] Any ideas on an Interactive IVR?
Hi!Does anyone have any ideas on how to build an interactive IVR where questions are asked by Asterisk (pre-recorded prompts), the caller answers the questions, and the system records the answers and emails the whole question-answer session as a .wav file? Similar to Comedian Mail except an menu would be required to play back each answer to see if the user would like to re-record, etc.Thanks!chris___ 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] Unable to make EM Wink work with T400P
Hi! I'm having a problem getting
a channelized T1 to signal correctly with a Nortel DMS 500. I've
got a 2-way trunk group set up with EM Wink start connected to Asterisk
on channel 1 of a T400P card. I'm pretty sure the configs are correct
(listed below) but when the Northern winks to Asterisk, Asterisk is not
returning the wink. Has anyone set up this configuration who could
possibly see an error on my part?
Thanks!
chris
__
zaptel.conf
span=1,1,0,esf,b8zs
span=2,1,0,esf,b8zs
span=3,1,0,esf,b8zs
span=4,1,0,esf,b8zs
em=1-24
loadzone=us
defaultzone=us
__
zapata.conf
[channels]
context=tester
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
tranfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
ragain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
signalling=em_w
group=1
channel=1-24
__
extensions.conf
[tester]
exten = _1NXXNXX,1,Dial(Zap/g1/${EXTEN:1},60,)
exten = _1NXXNXX,102,Busy
exten = _1NXXNXX,2,Congestion___
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[Asterisk-Users] How would you handle a fax without T.38 or G.711uLaw?
Let's say you were wanted to terminate calls onto your Asterisk system but your only available codec was G.729 and you had no control over the remote SIP proxy sending you the traffic. What would you do? Does anyone have an update on Asterisk supporting T.38 with SIP? Thanks! chris___ 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
[Asterisk-Users] Net2Phone, Asterisk, and 404 Not Found
Hi! Net2Phone is getting a common SIP status code, 404 Not Found, when trying to place a call to our Asterisk server. We're hoping someone on the list can shed some light on why this is happening. We can process a call from Asterisk to Net2Phone without any problems. Net2Phone sends the INVITE but immediately gets the 404 Not Found. The To: field of the INVITE contains the E.164 formatted number with a plus + sign before the 11 digits and we were thinking that the presence of that plus sign had something to do with the 404 problem. But I guess the plus sign is part of the SIP standard. I don't think we've seen the INVITE but I'll dig further on that. Has anyone connected Asterisk to a different SIP proxy and used SIP to communicate between the two? Can anyone further explain why our Asterisk is not replying to Net2Phone's INVITE? Here is the entry from our sip.conf file: [net2phone3] context = n2p-in host=Net2Phone's IP Address disallow=g723.1 allow=g729 type=friend dtmfmode=rfc2833 Thanks in advance! chris___ 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
Re: [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?
The Adtran Atlas 550 or 830 may do what you're looking to do. We use it to split PRI into multiple BRI. chris Nate Carlson [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 08/10/2004 01:30 PM Please respond to [EMAIL PROTECTED] To [EMAIL PROTECTED] cc Subject [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's? Hey all, I've got a PRI used for data calls right now, terminated in a MAX 4000. We're only using around 12 channels on average and 16 max, so I'd like to split off the remaining channels to terminate in an Asterisk box. Does anyone know of a device that'll take a PRI in, and spit out two PRI's that share the channels? Also need to be able to do some sort of call routing so that calls coming into the data number get terminated in our Max and the rest of the calls go to Asterisk. It looks like the following device will do it: http://www.isdnconnect.com/giff_pages/diagramb.htm ..but it's also rather expensive. (Looks like US ~$3k for one that supports 3 PRI lines.) We do also have a dual-PRI license for our MAX, so I'm investigating if it's possible to have it do the routing. I'd prefer to avoid running the data side of the PRI through Asterisk, as I'll be in big trouble if that stops working. :) Thanks! | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | |depriving some poor village of its idiot since 1981 | ___ 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
Re: [Asterisk-Users] Sipura line 1 outgoing voice problem?
__ Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has had this problem. For the meantime, I switched the internal phones so that my wife's favorite phone is line 2 and I told her to not pick up with line 1. Not a very permanent solution :) NAT is not an issue as the Sipura and * are on the same network. Is anyone else having this problem? It looks like other people are using Sipura (I saw one user with 30 of them ?!) and am surprised that nobody else is complaining about this problem. I am willing to step through some sip debug if anyone is interested in the output. * version: Asterisk CVS-02/08/04-22:22:57 Sipura firmware: 1.0.31 (just upgraded tonight to see if the problem would go away) Relevent config sections: --8-- sip.conf --8-- [cordless1] type=friend username=cordless1 secret=xxx host=dynamic context=cordless1 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw [cordless2] type=friend username=cordless2 secret=xxx host=dynamic context=cordless2 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw -- Chris ___ I had the exact same problem with a Mediatrix 1102doing a flash hook brought both sides of the conversation together. I found out that my sip.conf file had GSM as the first priority codec and the 1102 doesn't support GSM. I kept that the same but put a disallow = gsm statement in my sip entry for the 1102 so g.711ulaw would be the first negotiated codec. That fixed the problem. VZ
[Asterisk-Users] Do I have a bad T100P?
Has anyone encountered this issue: I'm bringing up a T100P card and ran 'modprobe wct1xxp' but Zaptel Tool shows the card as UNCONFIGURED in its alarm window. the LED on the card itself is not lit and not flashing. lsmod shows the module wct1xxp as zaptel both loaded. Thanks in advance for any advice from all of you Linux/Asterisk gurus! chris Chris Veazey Applications Engineer Black Hills FiberCom 605.721.2064 [EMAIL PROTECTED]
[Asterisk-Users] adtran 750 + t100p
__ Hi, I've had some issues with the x100p in my * box with echo at the beginning of calls and remote hangup detection. Question is, will setting up an adtran for the pots lines and connecting this to a t100p in * fix these issues ? Can anyone verify good call quality and remote hangup detection on incoming / outgoing US pots lines with the 750 + t100p combination ? Thanks, Chris Clifton Chris, I've found the Adtran 750 works very well with the T100P. I haven't noticed any echo on the lines. All the FXS lines from the Central Office are feeding FXO ports on the 750. The CO lines all have cutoff on disconnect enabled (Your phone company may call it something different. Basically, its a zero voltage across tip/ring for a few milliseconds). To take advantage of COD, your zapata.conf should be using fxs_ks (kewlstart) signalling. Chris Vz [EMAIL PROTECTED] ___
[Asterisk-Users] wct1xxp module and the T100P
I'm having trouble turning up a PRI to a T100P. I've read on the Digium FAQ's that once the wct1xxp module is loaded correctly, the LED on the T100P will flash red. I believe I've loaded the module correctly because both wct1xxp and zaptel are listed when I do the lsmod command. The LED on the card does not flash on and off. Does anyone have any recommendations on what I could be doing wrong? Thanks in advance! chris
Re: [Asterisk-Users] wct1xxp module and the T100P
Steven Critchfield [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 03/03/2004 04:06 PM Please respond to asterisk-users To:[EMAIL PROTECTED] cc: Subject:Re: [Asterisk-Users] wct1xxp module and the T100P On Wed, 2004-03-03 at 17:01, [EMAIL PROTECTED] wrote: I'm having trouble turning up a PRI to a T100P. I've read on the Digium FAQ's that once the wct1xxp module is loaded correctly, the LED on the T100P will flash red. I believe I've loaded the module correctly because both wct1xxp and zaptel are listed when I do the lsmod command. The LED on the card does not flash on and off. Does anyone have any recommendations on what I could be doing wrong? Could be wrong or bad cable, could be incorrect configuration, could be no service on the line yet. Must provide data to get information. -- Steven Critchfield [EMAIL PROTECTED] __ I know the PRI is good to the RJ-45 right before it goes into the T100P; I've made calls on it with our T-Berd. So I'm sure the cable is ok and there is service on the line. I haven't run Asterisk yet; shouldn't the card look alive once its driver module is loaded? Thanks! chris
Re: [Asterisk-Users] wct1xxp module and the T100P
Andrew McRory [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 03/03/2004 04:11 PM Please respond to asterisk-users To:[EMAIL PROTECTED] cc: Subject:Re: [Asterisk-Users] wct1xxp module and the T100P On Wed, 3 Mar 2004 [EMAIL PROTECTED] wrote: I'm having trouble turning up a PRI to a T100P. I've read on the Digium FAQ's that once the wct1xxp module is loaded correctly, the LED on the T100P will flash red. I believe I've loaded the module correctly because both wct1xxp and zaptel are listed when I do the lsmod command. The LED on the card does not flash on and off. Does anyone have any recommendations on what I could be doing wrong? Switch type, line code, framing all matter. How about posting your config? -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. PO BOX 3791 Tallahassee, FL 32315 (850)224-5737 (850)294-7567 ___ Sure: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us zapata.conf: [channels] context=cyperpri switchtype=national pridialplan=unknown signalling=pri_cpe channel=1-23 Here's a question, though: does the wct1xxp module read from either zaptel.conf or zapata.conf when loaded? Thanks! chris
Re: [Asterisk-Users] wct1xxp module and the T100P
On Wed, 2004-03-03 at 17:01, [EMAIL PROTECTED] wrote: I'm having trouble turning up a PRI to a T100P. I've read on the Digium FAQ's that once the wct1xxp module is loaded correctly, the LED on the T100P will flash red. I believe I've loaded the module correctly because both wct1xxp and zaptel are listed when I do the lsmod command. The LED on the card does not flash on and off. Does anyone have any recommendations on what I could be doing wrong? Thanks! chris Could be wrong or bad cable, could be incorrect configuration, could be no service on the line yet. Must provide data to get information. -- Steven Critchfield [EMAIL PROTECTED] __ I know the PRI is good to the RJ-45 right before it goes into the T100P; I've made calls on it with our T-Berd. So I'm sure the cable is ok and there is service on the line. I haven't run Asterisk yet; shouldn't the card look alive once its driver module is loaded? Thanks! Chris Nope, What would answer the request placed on the line. BTW, please see about putting some sane configuration options in your mailer. I know Lotus is notoriously crappy when viewed by other readers, but that was really bad. -- Steven Critchfield [EMAIL PROTECTED] __ Steven, Perhaps I should have posted my question differently to the list: After installing the CVS version of Asterisk, I type, modprobe xct1xxp. The machine accepts the command but the LED on the T100P does not flash. How do I know that the T100P module has loaded correctly? (What sort of mailer config options do you recommend?) Thanks! chris
[Asterisk-Users] Mediatrix 1102 issue after upgrading to CVS
We just did an upgrade on our Asterisk to the CVS version and our Mediatrix 1102s stopped working correctly. Our Asterisk is connected to the PSTN with a PRI. Calls from the PSTN to the Mediatrix 1102 work fine. The issue is calling out to the PSTN from the 1102. Asterisk looks like it process the call just fine except there is no talk path. Get this, though: If you flash hook and then flash hook again, the call is there. These units worked fine before our Asterisk upgrade. VoIP to VoIP works fine; it's just VoIP to the PSTN. Has anyone experienced something like this with the CVS version? (On the flip side, my Zultys Zip4x4 is now completely working. Before our upgrade, Zultys tech support said they had to create a software patch to overcome Asterisk leaving the contact field blank. I failed to load the patch before our upgrade and now that issue is gone.) Thanks in advance for taking the time to read this! chris
[Asterisk-Users] Re: Mediatrix 1102 issue after upgrading to CVS
__ We just did an upgrade on our Asterisk to the CVS version and our Mediatrix 1102s stopped working correctly. Our Asterisk is connected to the PSTN with a PRI. Calls from the PSTN to the Mediatrix 1102 work fine. The issue is calling out to the PSTN from the 1102. Asterisk looks like it process the call just fine except there is no talk path. Get this, though: If you flash hook and then flash hook again, the call is there. These units worked fine before our Asterisk upgrade. VoIP to VoIP works fine; it's just VoIP to the PSTN. Has anyone experienced something like this with the CVS version? (On the flip side, my Zultys Zip4x4 is now completely working. Before our upgrade, Zultys tech support said they had to create a software patch to overcome Asterisk leaving the contact field blank. I failed to load the patch before our upgrade and now that issue is gone.) Thanks in advance for taking the time to read this! chris I'm sorry; I think I found the issue. We've been playing with the codec preferences and had GSM before g.711mulaw. Since the Mediatrix 1102 doesn't have a GSM codec, I think it had trouble establishing a codec link. By doing a flash-hook twice, that was enough to get the call through. SO, I don't think it had anything to do with the CVS upgrade. I changed the order of codec preference and calls are going through just fine. Thanks! chris
RE: [Asterisk-Users] PRI D Channel and Caller-ID issue......
Adams, Gavin [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 01/08/2004 06:55 AM Please respond to asterisk-users To:[EMAIL PROTECTED] cc: Subject:RE: [Asterisk-Users] PRI D Channel and Caller-ID issue.. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ryan Tucker On Wed, 7 Jan 2004 [EMAIL PROTECTED] wrote: delivered in the D Channel setup message. If a call comes to our switch from off-network, i.e. the LEC, long distance, or a cellular provider, only the caller number is sent in the setup message. We've found out that the caller name information is sent but in a later D channel message, not the setup. Is there a way to have Asterisk gather that information from subsequent messages? Hmm... we have a similar situation. Where do you need the data? I've noticed that it does end up there in time for voicemail notifications and writing the CDR out, but not in time to ring SIP endpoints. Does it show up on a show channel? -rt Mark answered this a couple days ago for me. Do a pri debug span and see if the caller name is in a FACILITY message. If so, Steve's comment about adding a Wait(1) as the first priority for the incoming call will work. Regards, --- Gavin ___ Gavin, Steve, and Ryan, I wanted to thank you; that 1 second delay on the incoming call fixed the issue. I did the PRI debug like you suggested and found that the D-channel adds a FACILITY message type (with caller name included) if the call is from outside our switch. If the call is inside the network, the D Channel puts the caller name in the SETUP message. It must have something to do with off-net calls dipping to remote databases for that information. We have our switch set to not dip to SS7 if the call is inside our switch. Again, thanks. Chalk this up as another satisfied Asterisk list user! chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI D Channel and Caller-ID issue......
I was wondering if anyone has encountered and overcome this situation: We've got a PRI to our Asterisk system and notice that if a call comes in from a phone on our network, both caller name and caller number are delivered in the D Channel setup message. If a call comes to our switch from off-network, i.e. the LEC, long distance, or a cellular provider, only the caller number is sent in the setup message. We've found out that the caller name information is sent but in a later D channel message, not the setup. Is there a way to have Asterisk gather that information from subsequent messages? Thanks for input anyone can provide! chris
[Asterisk-Users] Mediatrix 1102 / 1104 authentication problems....
Hi! Has anyone on the board successfully installed a Mediatrix 1102 or 1104 as a SIP peer on Asterisk? I'm trying to configure different user accounts on each FXS port, but I'm having authentication problems; Asterisk is saying the client is not authorized. Interestingly enough, I can dial a 9 and make a local call through the Mediatrix. Thanks! chris
[Asterisk-Users] Asterisk with External Voicemail
If anyone could help me with this, I'd appreciate it!
I've got an Asterisk deployment where I'd like to use an existing external Octel voicemail system. I've been trying to define an extension that if the call isn't answered in a few rings, to dial our external voicemail number. That voicemail system works by seeing the CALLED number and routing the call to the user's answering message. The problem is that Asterisk sends the CALLING digits on the outbound call and the CALLING party gets the welcome to Octel message instead of the user's voicemail box.
Is there any way to a variable like ${EXTEN} or ${RDNIS} to replace the caller ID information with the destination mail box number?
Am I thinking through this the wrong way?
Thanks!
Chris Veazey
