[Asterisk-Users] sip.conf user entry for ViaTalk
Try as I might, I can not get incoming calls from ViaTalk to match against my user entry. I have both peer and user entries, and incoming and outgoing calls work, but incoming calls do not move to my in-viatalk context (they stay in the default context.) Has anyone else managed to get this to work? My user entry looks like: [viatalk-in] username=1407965 context=viatalk-in type=user host=965.407.1.switch.vtnoc.net I've also tried username=+1407965, host=67.15.74.73, host=67.15.74.73:5060, and host=dynamic. SIP debug from an incoming call shows: <-- SIP read from 67.15.74.73:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 67.15.74.73:5060;branch=z9hG4bK6169ed4e;rport From: "Wern Ben" ;tag=as7366fb31 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 18 Aug 2005 03:48:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 214 charon*CLI> v=0 o=root 16334 16334 IN IP4 67.15.74.73 s=session c=IN IP4 67.15.74.73 t=0 0 m=audio 21762 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - The next message indicates that it can't find a user or peer to match "67.15.74.73:5060", and moves to the default context. In the above examples, I've 'd the last four numbers of live phone numbers. ViaTalk appears to be sending the incoming caller info (including plus sign) in the From: part, and not my userid. Does anyone have any suggestions? Ben Wern ___ 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] too many ex-(boy|girl)friends
That's pretty good..
I have a similar situation, where I need to match all the area codes in
a particular state like:
exten => _[904|321|407|252]XXX,1,Dial..
But it doesn't work. I can get it to work with something along the lines of:
exten -> _[904|321|407|352]X.,1,Dial
But I was hoping to be more specific.. other than specifying each area
code ala _904XXX,1,Dial. do you know of any way to do this?
Ben Wern
> Maybe like this:
>
>
> exten => s,5,DBGet(blacklisted=blacklisted/${CALLERIDNUM}) exten =>
> s,6,GotoIf(${blacklisted} = "1"?hell|1)
>
> You just have to put every blacklisted number in the Asterisk database as
> it would be seen from the callerid number.
>
> I this this solution is better than changing your extensions.conf every
> time you change (boy|girl)friend.
>
> Michel
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Re: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting
Chris, >You must have call waiting turned off on your comm pilot control panel, I didnt even have that option in my "Comm Pilot" web interface; after working with Broadvoice support further, they determined that the account had not been fully provisioned -- something went south half-way through. Ben Wern ___ 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] Broadvoice BYOD Plans - 3-way and Call Waiting
Jeff, >I believe the whole issue in general has something to do with BroadVoice >not setting the privacy bit in the SDP for the call, indicating an >anonomous caller id. As such, it's taking whatever it can for the >caller ID, which happens to be the IP of the server that sent it the >call. Thanks; I forward this to Broadvoice last week, and they appear to have corrected this issue -- I'm now getting the full CID information. Not sure if they did this in particular or something else, but thanks for your suggestion! Ben Wern ___ 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] Sprint PCS -> Asterisk through Digium TDM400P
Unfortunatly, I can't offer any suggestions to correct the issue, but I can add that I see this on my X100P as well. It does seem particularly bad with Sprint; I think they may have shorter tone lengths than we really need. The only thing that I've found so far is that my calls coming in via SIP and IAX do not exhibit the problem, so I've transitioned the bulk of my incoming traffic to those mechanisms -- may not be an option for you of course. Ben Wern -- Original Message --- From: "Alok K. Dhir" <[EMAIL PROTECTED]> To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thu, 16 Sep 2004 15:32:52 -0400 Subject: [Asterisk-Users] Sprint PCS -> Asterisk through Digium TDM400P > Does anyone have trouble with dialing in to an Asterisk Server and > having the DTMF digits recognized? We have some clients who are > calling in with cell phones, notably those with SprintPCS service, > who's DTMF is just never recognized. > > I have tried relax_dtmf on and off, with no improvement. My rxgain > is currently set to 3. > > Can anyone suggest possible solutions? > > Incoming calls are coming through POTS lines connected to the server > to TDM400P with FXO modules. > > Thanks, > > Al > > -- > Alok K. Dhir <[EMAIL PROTECTED]> > Symplicity Corporation > http://solutions.symplicity.com > 703 351 6987 (w) | 703 351-6357 (f) > ___ > 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 --- End of Original Message --- ___ 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] Broadvoice BYOD Plans - 3-way and Call Waiting
I've already run into some trouble with Broadvoice. Broadvoice support tells me that support isn't really available to BYOD plans, which I suppose I understand given the variety of devices out there. I'm hoping that someone on Asterisk-Users has seen the two issues I'm running into and has a suggestion. The first issue I'm seeing is that incoming caller id shows the number as "out of area" and the name shows as "147.135.8.129;bvoice" I don't have this problem with other incoming SIP providers -- is there some tweak I need to make Asterisk see CID information from Broadvoice? The other issue is that call waiting does not appear to work. The way I'm expecting it to work with Asterisk is to send the second call to me - I'm using SetGroup and CheckGroup within Asterisk to limit my calls to two at a time total. However, if I'm on a phone call (incoming or outgoing), Broadvoice transfers a second call to a "person you are calling is busy" message -- I don't see any additional SIP traffic to the Asterisk box. Ben Wern ___ 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] Broadvoice BYOD Plans - 3-way and Call Waiting
To followup, I did finally get a response from Broadvoice indicating that two simultaneous calls are allowable on BYOD plans, which would allow Asterisk to handle the three-way and call waiting functions. I've just signed up to verify this. Ben Wern ___ 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] Broadvoice BYOD Plans - 3-way and Call Waiting
Kevin, Thanks; from what I have read on other Broadvoice threads, that has to do with comfort noise generation.. more of an asterisk issue than anything else. As a followup, I did get a response from broadvoice after posting to this forum indicating that they are checking with ther billing department. I will update this thread when I get a response. Ben Wern -- Original Message --- From: Kevin <[EMAIL PROTECTED]> To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Sun, 29 Aug 2004 21:30:58 -0400 Subject: RE: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting > I have been unable to get the asterisk voicemail to work reliably > with broadvoice. ___ 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] Broadvoice BYOD Plans - 3-way and Call Waiting
Can anyone who is using Asterisk with Broadvoice tell of their experiences with 3-way calling and call waiting? I can't get Broadvoice to respond to my question, but I understand that there is a per minute fee (3.9 c/minute?) if you go over your use allowances. My question is, how are 3 way and call waiting calls handled? Because Asterisk would just handle them as two different channels/calls -- does Broadvoice allow BYOD customers to have two active lines and then start charging for a third? If so, does anyone have any configuration examples of limiting the number of sessions to a single provider? Ben Wern ___ 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] Broadvoice BYOD Plans
Can anyone who is ___ 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] call waiting, * and FXO
Not in any way a good solution, but what I've done is create an extension that flashs the line, and then returns the call to my sip phone. For example:
[app-flash]
exten => _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM})
[macro-test]
exten => s,1,Answer
exten => s,3,Flash
exten => s,3,Dial(SIP/${ARG2},30,t)
exten => s,4,Dial(SIP/${ARG1},30,t)
exten => s,t,Hangup
exten => s,i,Hangup
exten => s,h,Hangup
Then if you're on a call through the Zap line, and transfer the call to *4, it will flash the line and return it to SIP extension. I've been trying to get it to auto-detect the SIP extension to return it to, but callerid is different depending on if the call is incoming or outgoing through the Zap.
Again, not good.. but works in a home environment. I think we'll need in-call triggers to do anything better.
Ben Wern
-- Original Message ---
From: "mike jennings" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wed, 28 Jul 2004 22:38:41 -0500
Subject: [Asterisk-Users] call waiting, * and FXO
> I have been told that the combination of call waiting, * and FXO does and will not work because Asterisk is a PBX. I guess Id like to hear if this is a hard and fast no this will not work and heres why, or that this currently doesnt work but with some coding might work.
>
> Id like to have the option to be able to continue using call waiting with an FXO line (and I know Im not alone). I know if I switched to a SIP based connection instead of the FXO this would work, but I currently like my unlimited plan with Vonage.
>
> Would anyone like to enlighten me?
>
> I have done numerous searches and Ive included a few postings that were mostly not answered.
>
> http://lists.digium.com/pipermail/asterisk-users/2004-May/046855.html
> http://www.vovida.org/pipermail/mgcp/2001-May/000571.html
>
> Thanks
--- End of Original Message ---
[Asterisk-Users] X100P Call Waiting and Three Way Calling from SIP Device
I'm trying to be able to access the call waiting and three-way calls
features on a line attached to my X100P. For example, a party calls, the
X100P/Asterisk ring the 7960 on my desk, and all is fine. If I want to three
way call another individual in, I need to send a Flash to the X100P, and the
7960 doesn't appear to have any way to to that mid-call. All I can come up
with is transferring the call to a macro that will Flash, Dial the digits,
and return the call to me. For example, _*4. points to:
[app-flash]
exten => _*4.,1,Flash()
exten => _*4.,2,SendDTMF(${EXTEN:2})
exten => _*4.,3,Flash()
exten => _*4.,4,Transfer(1112)
This seems to work.. almost.. The flash, DTMF, and Flash commands work,
becuase the party on the first Zap call can hear the party on the second Zap
call. However, the Transfer back to the 7960 doesn't work, and after a few
seconds the entire call is dropped. Any idea on what I'm doing wrong? Is
there a simpler (in-call flash?) way to do this?
Ben Wern
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[Asterisk-Users] Residential Plans for Asterisk Users
Does anyone have any suggestions on companies that offer bundles ala Vonage or Voicepulse for Asterisk users? There are a number of great vendors for per-minute plans, but since I use asterisk at my home, I'd like to do so through an "unlimited long distance" plan. Does anyone have any reccomendations? Ben Wern ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Manually Answer
>[inbound-home]
>exten => s,1,Dial(${PHONE3}&${PHONE4},15)
Thanks Rich, this worked like a charm, I don't know why I was thinking in
reverse -- that I would have to have Asterisk answer it to pass the ringing
to the SIP phone or I would have to "force" a pickup with some sort of key
sequence.
Thanks!
Ben Wern
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[Asterisk-Users] X100P Manually Answer
I have an X100P used, at present, largely for outgoing calls. It shares the single incoming POTS line with a number of analog phones. Is it possible to talk the X100P (Zap/1) to answer a ringing call only if I ask it to? I'd like to use only the SIP phone in my office, but let the analog phones continue to work in the rest of the house (until I can afford FXS cards anyway..) I can force it straight to the POTS line with something like: exten => 44,1,Dial(Zap/1/,10,t) but it won't attempt the dial / pick up the phone if it's ringing. Thanks, Ben Wern ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Follow Me
Ernest, Again, I really appreciate your help with this. Your solution looks like it requires two POTS lines -- am I misreading it? My goal is to have a call come in on a single POTS line and then have Asterisk try to track me down via the same POTS line (3 way calling.) Ben At 12:30 PM 9/17/2003 -0700, Ernest W. Lessenger wrote: At 06:48 PM 9/16/2003, you wrote: cell phone into the call (or my office number, etc.) I understand the selected numbers part of it, but not how to get it to use the three way. If I send it to Nufone first, I'm paying for a call to a local number (my cell) that I don't need to. This should work... [default] exten => s,1,Dial(Zap/3,20,t) ; This is your desk phone exten => s,2,Dial(Zap/2/1234567,20,t) ; This is your secondary POTS line calling your office exten => s,3,Dial(Zap/2/3217654,20,t) ; This is your secondary POTS line calling your cell phone ; I've never tried this one coming up, but I think it's worth a shot as it works just fine for local extensions exten => s,4,Dial(Zap/2/3217654&Zap/3/3217654,20,t) ; This is your secondary and tertiary POTS lines calling your cell phone anbd office As long as none of these lines go to voicemail, they should fail over properly in order. You can also make it more complicated with time-based includes and gotos. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Follow Me
Ernest,
I hadn't thought of doing that, though having that added protection would
be nice. However, what I'm trying to do it have an incoming call at my home
number follow me to my cell phone for selected numbers -- Since I already
have three way calling, I'd like get Asterisk to essentially three way my
cell phone into the call (or my office number, etc.) I understand the
selected numbers part of it, but not how to get it to use the three way. If
I send it to Nufone first, I'm paying for a call to a local number (my
cell) that I don't need to.
Ben
At 09:57 AM 9/16/2003 -0700, Ernest W. Lessenger wrote:
At 11:22 PM 9/14/2003, you wrote:
First -- Thanks to everyone who offered their help and tips on getting my
Cisco 7960 working with Asterisk -- this is great stuff.
Does anyone have any examples of "Follow Me" or other call forwarding with
a single PSTN interface? Or a pointer on what I need to read to figure it
out?
Is this what you need? Basically, the local trunk and the Nufone trunk
fail over to each other. So, if you have a forward set up and transfer to
a non-local extension, the call will go out even if the original incoming
call was made on the PSTN line.
[trunklocal]
exten => _NXX,1,Dial(${TRUNK}/${EXTEN})
exten => _NXX,102,Dial(${NUFONE}/1${AREACODE}${EXTEN})
exten => _NXX,203,Congestion()
[iaxprovider]
exten => _1NXXNXX,1,Dial(${NUFONE}/${EXTEN})
exten => _1NXXNXX,102,Dial(${TRUNK})
exten => _1NXXNXX,203,Congestion()
exten => _011.,1,Dial(${NUFONE}/${EXTEN})
exten => _011.,102,Congestion()
exten => _1011.,1,Dial(${NUFONE}/${EXTEN})
exten => _1011.,102,Congestion()
--Ernest
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[Asterisk-Users] Follow Me
First -- Thanks to everyone who offered their help and tips on getting my Cisco 7960 working with Asterisk -- this is great stuff. Does anyone have any examples of "Follow Me" or other call forwarding with a single PSTN interface? Or a pointer on what I need to read to figure it out? Thanks, Ben Wern ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960
Andrew, Thanks for your help! I did have the outgoing proxy set -- since I had FWD set up on line 1. I removed all the FWD stuff, and the outgoing proxy. I altered the entry to have the qualify, canreinvite, and nat lines and also altered the user id to be a number. Now I'm able to call other local extensions, but I can't call into the Cisco. But it's progress! I can also call out to FWD, but audio drops after a few seconds. Don't even want to think about getting FWD calls back into the network. exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) This didn't work - what does the @1000 indicate? Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco 7960
Andrew, I removed that entry, still no luck. I also altered the config to use a number (101) as the entry name instead. I get: Got SIP response 404 "Not Found" back from 172.16.1.28 SIP/101-e9a4 is circuit-busy on the console when I try to call it. sip show peers shows the node, with an OK status. Ben At 02:09 AM 8/30/2003 -0400, Andrew Joakimsen wrote: > My sip.conf entry for the cisco looks like this: > > [cisco] > type=friend > username=cisco > secret=1234 > host=dynamic > defaultip=[The IP of the 7960] > mailbox= > context=sip > callerid="Ben" <1> Try to remove the defaultip= string. Do you get any errors in the console when it is run in verbose mode? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Cisco 7960
I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent "Getting Started with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco secret=1234 host=dynamic defaultip=[The IP of the 7960] mailbox= context=sip callerid="Ben" <1> And the related extensions.conf entry: exten => 1,1,Dial(SIP/cisco,20,tr) The Cisco config itself.. Line 1 is set for FWD. Line 2 is: Name: cisco Shortname: cisco Authentication Name: cisco Authentication Password: 1234 Display Name: cisco proxy address: [The IP of my Asterisk installation] proxy port: 5060 The FWD line works, the Asterisk line doesn't. Any suggestions or pointers to documentation I might have missed? Thanks, Ben Wern ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
