[asterisk-users] SIP Source Port
Hi All. We have a provider that requires us to SOURCE the SIP connection on TCP 5061. I honestly have no clue how to force Asterisk to always SOURCE the SIP connection on a certain port. Can anybody point me in the right direction? I am using PJSIP. Thank you, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STIR/SHAKEN
Hi All. The folks at TILTX have set up a Facebook Live event for Wednesday, May 26, 2021 at 12:00 PM Eastern Time. According to TILTX, this will cover STIR/SHAKEN and how Asterisk works with it. If anybody is interested, here is the link. https://www.facebook.com/events/489246355856999/ Thanks, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup Reason
Hi All. We've put in a check for Do Not Call before a call goes out. However, we have noticed that we cannot seem to pass a 'hangup reason' for a call. For example, I'd like to know that this number is on the DNC so our system does not call them back. Is it possible to pass a hangup reason to Asterisk? Not so much a code, but a reason. Or is there a code for DNC? Thanks all, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STIR/SHAKEN
Hey All. I spoke to the guys at TILTX and they agreed to host a 30 minute webinar for STIR/SHAKEN and Asterisk. They will coordinate internally and they will send me an invite. I will share this invite in the event anybody would like to join. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STIR/SHAKEN
Hi Jeff. What exactly do you mean by the 'inbound piece'? I've spent quite a lot of time with the folks at TILTX understanding the framework; but I am not exactly sure what you mean by the 'inbound piece. Greg/Doug, like many folks here, we use LCR. So, the terminating carrier is not necessarily the one that issued us the telephone numbers. So, they will not sign it or simply cannot sign it. Remember that a very limited number of companies can actually sign the calls; the rest have to buy it from these 'Service Providers'. And there is another situation - the company you purchase your numbers from and the company you place your calls through may be different and both may not be able to sign your calls. Again, a very limited number of service providers that can actually sign your calls. So what do you do in that scenario? You have to find a Service Provider that can: 1. Verify you own that telephone number(s). 2. Sign your calls. 3. Provide you with the technical means to do so. So, that's that... I hope this makes sense. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STIR/SHAKEN
Hi Greg. In our use case, we purchase DIDs from them. So, they are the inbound carrier (they are a CLEC and IPES) and STIR/SHAKEN Service Provider. However, we do not use them for termination. They offer service termination, but we do not use them due to existing contracts. So, in order to have our calls signed, we needed them to do it. The biggest issue we've come across is the number of companies *able to *provide this service is limited, especially to the Asterisk community. I stress able to because even though some companies are Service Providers, they are simply not technically capable of offering it. I will send you my contact's information at TILTX privately. He's a subject-matter expert with the STIR/SHAKEN framework and he's offered us invaluable help. Thanks, Alex On Sun, Mar 7, 2021 at 1:43 PM Greg Troxel wrote: > > Alexander Perkins writes: > > > They ended up creating an AGI script for us that handles everything. At > > the end of the day, all we needed to do was pull down the script, and add > > the exten => s,n,AGI(TILTX-SHAKEN.agi) command and it handles > > everything else. > > I wonder if you could step back and explain the big picture, as I'm not > really following this. As I understand it: > > usually asterisk is used as a pbx > > STIR/SHAKEN is a protocol run between carriers to prove the authority > to use the claimed callerid > > when someone gets service from a carrier and connects to it from > asterisk, I would expect the carrier to basically filter the claimed > callerid to be from the set of values recorded with your account as > legit, and for the carrier to do the STIR/SHAKEN authentication. > > So I wonder if your asterisk instance is connecting to the PSTN as a > top-level carrier, or, more likely, I am confused in some way. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STIR/SHAKEN
Hi All. I wanted to give an update to this as we've been working closely with the Technology Innovation Lab (TILTX) and getting this working on our Asterisk boxes. They ended up creating an AGI script for us that handles everything. At the end of the day, all we needed to do was pull down the script, and add the exten => s,n,AGI(TILTX-SHAKEN.agi) command and it handles everything else. Anyways, I am sharing this because it took us a long time to find a STIR/SHAKEN Service Provider that would work with us. These guys not only worked with us, but they created something super-simple for us at no charge. Highly recommend them. Here's their email address for this - 0...@tiltx.com Hope this helps. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Script Returning 4
HI All. I have a really strange issue that I'm two months into troubleshooting; however, I cannot figure it out. I have an AGI Script (PHP) that runs every time a call comes into my Asterisk box. Most of the time, it runs without any issue. However, every now and then, the PHP-AGI script fails after it is executed and simply returns 'returning 4'. I verify the PHP script begins to run. However, it appears to just stop. I have placed try/catch statements everywhere, but it does not seem to hit them. Just to verify, this is the same script running over and over with the same parameter. Any ideas/suggestions as of what can be happening? Thanks, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STIR/SHAKEN
Jeff, yes. The process is long. It is actually around one year. We ended up going with a SHAKEN Service Provider named Technology Innovation Lab ( www.tiltx.com). They have been awesome. They are certified in Asterisk and catered the solution to our Asterisk install. Highly recommend them. Their email for SHAKEN is 0...@tiltx.com. Anyways, give them a shot. Took us a while to find a SHAKEN Service Provider that knew Asterisk. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP TLS, Not HTTPS
Hi All. We are trying to get SIP TLS working, but have run into a snag. We followed this documentation - https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial - but, when it comes to PJSIP, we are bit lost on the authentication process. For example, in that documentation, the peer has assigned a username and password. However, in our case, we are simply doing IP-based authentication. I've tried Googling and all that, but I am not coming across an answer (I am probably not searching correctly). Anyways, how would I go about using an IP address to authenticate the peer for TLS, not a username/password? Thank you, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get a SHAKEN Identity Token
Hi Markus. Thanks a bunch! I will try that out!
On Fri, Jan 22, 2021 at 8:06 AM Markus wrote:
> Am 07.01.2021 um 23:49 schrieb Alexander Perkins:
> > Hi All. We have old Asterisk servers, 1,89, (we cannot upgrade because
> > of several reasons) and we are now implementing SHAKEN via our
> > provider. We place a SIP call to our provider and they return a 302
> > (information below). I am trying to get the X-Identity information
> > below, but I do not seem to be able to do so. Can somebody help me with
> > this? Any suggestions on how to get it?
>
> I use SIP_HEADER to extract information from inbound SIP packets and
> SIPAddHeader to copy that info to the outgoing call leg. Maybe this
> helps you?
>
> Example:
>
> exten => _+X.,1,NoOp(${CALLERID(num)})
> exten => _+X.,n,Set(PAI=${SIP_HEADER(P-Asserted-Identity)})
> exten => _+X.,n,Set(PAI=${CUT(PAI,:,2)})
> exten => _+X.,n,Set(PAI=${CUT(PAI,@,1)})
> exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "anonymous"]?anonymous:cli)
> exten => _+X.,n(anonymous),SIPAddHeader(P-Asserted-Identity: "${PAI}"
> )
> exten => _+X.,n,SIPAddHeader(Privacy: user\;id)
> exten => _+X.,n,Goto(dial)
> exten => _+X.,n(cli),SIPAddHeader(P-Asserted-Identity: "${PAI}"
> )
> exten => _+X.,n,SIPAddHeader(Privacy: id)
> exten => _+X.,n,Goto(dial)
>
>
>
--
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[asterisk-users] Get a SHAKEN Identity Token
Hi All. We have old Asterisk servers, 1,89, (we cannot upgrade because of several reasons) and we are now implementing SHAKEN via our provider. We place a SIP call to our provider and they return a 302 (information below). I am trying to get the X-Identity information below, but I do not seem to be able to do so. Can somebody help me with this? Any suggestions on how to get it? Thank you, All. Very much appreciated! <--- SIP read from UDP:XXX.XXX.XXX.XXX:5060 ---> SIP/2.0 302 STIR/SHAKEN Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5066;received=XXX.XXX.XXX.XXX;branch=z9hG4bK37b49c97;rport=5066 From: "12125551212" ;tag=as0026c4e3 To: ;tag=bcaa-20103108495689bb4065d39c43badb69 Call-ID: 6755a4484427f12b0e56d6903fe50...@xxx.xxx.xxx.xxx:5066 CSeq: 102 INVITE X-Identity: eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9jZXJ0aWZpY2F0ZXMuY2xlYXJpcC5jb20vZmRmYjMzMjgtYjc1NC00YTBkLThiMzQtZGUzMGIwOGFkYWMyLzQ3NmMyODliYTgxY2QxNWU3MjBmNzkxOWM5NGU5MzU2LmNydCJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMjU2NTI3NjIwMSJdfSwiaWF0IjoxNjEwMDU2MDE3LCJvcmlnIjp7InRuIjoiMTI1NjkwNjQ5NTUifSwib3JpZ2lkIjoiMGFlODFjZWQtYzhlZS00ZWFiLTliNjAtMDY3OWM0Y2Q1MjUwIn0.lr3uj0fmlHbSori-msdbvKu5SQrVnLA-ZMswCY_dLk79jrpr1yFhWmL4GiAr16VtMKVSamQ-0bi3Pptoi7TUfw;info=< https://certificates.clearip.com/fdfb3328-b754-4a0d-8b34-de30b08adac2/476c289ba81cd15e720f7919c94e9356.crt >;alg=ES256;ppt=shaken Server: TILTX Technology Innovation Lab SHAKEN Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Block Spam Calls
Hi All. Thank you so much for all the feedback; it is really helpful. I found a company out there that specializes in Asterisk solutions and has a Robocall/SPAM call solution specifically for Asterisk. We give it a spin last week and it turned out great. It's not free, but works very well and I think it is worth passing it on. Here's the site for the service - https://www.tiltx.com/asterisk-robocall-blocker. I thought I'd pass this along as it has taken me some time to find something for Asterisk that actually works. Hope this helps, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Block Spam Calls
Hi All. Does anybody know if Google/Android has an API I can sign up for that will allow us to query the caller ID and find out if it is spam or a robocaller? I ask because we've had increase in spam calls and I'd like to simply play dead air or something really annoying. Thanks all, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic Agents Unavailable
Hi All. I had a situation today where all the dynamic agents became 'available'. This is a backup system we have that our folks call in from their cell phones and wait for calls while we fix the primary system. However, when we tested today, they all appeared as 'unavailable' and no calls that were in the queue wen to them (screenshot below). Is there a setting I can change to tell Asterisk to always make the agent available, unless the queue transferred them a call? Here's a screenshot of 'queue show': http://drops.tiltx.com/qR7nLD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Centralized Voicemail Server
Hi All. I have an interesting situation. I have two Asterisk servers and one Asterisk Voicemail Server, which serves voicemail to both boxes. My question is - how do I get the MWI to work for the end users since the two Asterisk servers do not have voicemail and the end-users are not registered to the voicemail server? Many thanks, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom BLF Question
Hi All. I have an interesting scenario. We use the Polycom VXX phones and have an auto-attendant on our Asterisk system. The receptionist can turn the auto-attendant off and on as she would like (she dials 444 to enable and 555 to disable). However, I’d like to have one of the BLFs on her Polycom light up if the auto-attendant is enabled and off if it is disabled. Any suggestions on how I can have the one of the Polycom BLFs stay on if the auto-attendant is enabled? Any help is appreciated Thanks, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Play Music While Processing AGI Script
Hi All. I have a question - I have an AGI script that may run for 10
seconds, or it may run for 60 seconds while an agent becomes available
(agents are geographically dispersed). Is there a way to have the music
play in the background while the AGI scripts executes? When the AGI script
finishes, then the music should also finish.
I tried this, but the music needs to finish before moving on to step 4 and
execute the script.
exten => _NXZNXX,1,Answer()
exten => _NXZNXX,2,MusicOnHold()
exten => _NXZNXX,3,AGI(SetRecordingID.php,${UNIQUEID})
Any help would be appreciated.
Alex
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