IRC log for #asterisk on 20210411

00:00.56*** join/#asterisk Janos (~textual@201.204.94.76)
00:40.45*** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir)
01:28.54*** join/#asterisk tsal (~tsal@i59F5F03A.versanet.de)
01:47.09*** join/#asterisk Edeneye (58a28967@88.162.137.103)
02:01.56EdeneyeSomeone using nexmo/vonage can pm me please
02:13.12SamotEdeneye: This is getting out of hand.
02:14.06SamotEdeneye: You really need to actually troubleshoot the configuration issue. You have now gone through at least three providers over the span of three days.
02:22.35SamotEdeneye: What is the actual issue you are having?
02:25.54*** join/#asterisk mr44er (~mr44er@x52716356.dyn.telefonica.de)
02:28.44SamotEdeneye: Well?
02:28.49EdeneyeSamot I can almost do everything. Like I told earlier I can now make calls, and I am under pjsip now. Everything works. Except that I can't change the caller ID. The number appearing on the calls I make is not even the number I bought from nexmo. I don't even now where this number is from
02:28.54EdeneyeI send you somthing
02:29.14SamotNo. In channel.
02:29.36EdeneyeYes here
02:29.38SamotShow your actual Dialplan for this.
02:29.39Edeneye[transport-udp]
02:29.40Edeneyetype=transport
02:29.40Edeneyeprotocol=udp
02:29.41Edeneyebind=0.0.0.0
02:29.41Edeneye[nexmo]
02:29.42Edeneyetype = endpoint
02:29.42Edeneyeaors = nexmo
02:29.43Edeneyeoutbound_auth = nexmo-auth
02:29.43SamotNo
02:29.43Edeneyecontext = from-yay
02:29.44Edeneyetransport=transport-udp
02:29.44Edeneyefrom_user = unknown
02:29.45Edeneyeallow=alaw
02:29.45Edeneyeallow=ulaw
02:29.45SamotStop
02:29.46Edeneyeallow=g729
02:29.46Edeneye[nexmo]
02:29.47Edeneyetype = aor
02:29.47Edeneyecontact = sip:sip.nexmo.com
02:29.48Edeneyequalify_frequency = 120
02:29.48SamotStop
02:29.53SamotJFC
02:29.59Edeneye; ----------------------
02:29.59Edeneye[6003]
02:30.00Edeneyetype=endpoint
02:30.00Edeneyeaors=6003
02:30.01Edeneyeauth=6003-auth
02:30.01Edeneyeallow=ulaw
02:30.02Edeneyecontext=to-yay
02:30.02Edeneyeice_support=no
02:30.03Edeneye;rtp_symmetric=yes
02:30.03Edeneye;force_rport=yes
02:30.04Edeneye;rewrite_contact=yes
02:30.04Edeneyedtmf_mode=rfc4733
02:30.05Edeneyedirect_media=no
02:30.05Edeneye[6003-auth]
02:30.06SamotDO NOT FLOOD THE CHANNEL
02:30.06Edeneyetype=auth
02:30.06Edeneyeauth_type=userpass
02:30.07Edeneyepassword=1234
02:30.07Edeneyeusername=6003
02:30.10SamotSTOP
02:30.19Edeneyebind=0.0.0.0
02:30.19Edeneye[nexmo]
02:30.20Edeneyetype = endpoint
02:30.20Edeneyeaors = nexmo
02:30.21Edeneyeoutbound_auth = nexmo-auth
02:30.21Edeneyecontext = from-yay
02:30.22Edeneyetransport=transport-udp
02:30.22Edeneyefrom_user = unknown
02:30.23Edeneyeallow=alaw
02:30.23SamotHoly hell
02:30.23Edeneyeallow=ulaw
02:30.24Edeneyeallow=g729
02:30.24Edeneye[nexmo]
02:30.25Edeneyetype = aor
02:30.25Edeneyecontact = sip:sip.nexmo.com
02:30.26Edeneyequalify_frequency = 120
02:30.26Edeneye[nexmo-auth]
02:30.27Edeneyetype = auth
02:30.27Edeneyeauth_type = userpass
02:30.58*** join/#asterisk jess (jess@freenode/staff/jess)
02:30.58SamotEdeneye: That was worthless.
02:31.11SamotEdeneye: use pastebin
02:32.09SamotAnd it wasn't even what I asked for.
02:33.25Samotjess: how funny that you showed up when the channel was being flooded. 🤔
02:34.02EdeneyeI told you before the flood. Sorry bout this I didnt realise it was that long
02:34.02jesssomeone told me there might be noise
02:34.29jess( 02:29:48 [freenode] !stross.freenode.net Possible Flooder Edeneye[58a28967@88.162.137.103] on stross.freenode.net target: #asterisk )
02:34.36SamotNothing major.
02:34.46SamotN00b
02:34.50jessthought i'd make sure everything was ok :)
02:35.21EdeneyeSo here it is
02:35.23Edeneyehttps://pastebin.com/3qZ9B2UP
02:36.38SamotSo that number isn't being set?
02:37.26EdeneyeNo
02:38.04EdeneyeThe point is to change the caller ID. As the one appearing on calls is not even the one from the provider
02:38.18SamotPerhaps it is because the variable is wrong.
02:38.18Edeneyeits some weird number like +133334444
02:38.36SamotCALLERID(num)
02:38.36EdeneyeWhich one ?
02:38.52SamotNot CALLERID(NUM)
02:38.53EdeneyeI check
02:39.04SamotTry that
02:41.09EdeneyeSo if this works in theory I can replace it by any caller ID i want
02:41.23SamotSure
02:42.08SamotIn Asterisk. Whether or not the providers accepts it, don't know.
02:42.59EdeneyeYes I know about this. Crossed fingers
02:47.24EdeneyeOk so I set exten => _X.,1,Set(CALLERID(NUM)=0675138545) as exten => _X.,1,Set(CALLERID(num)=0675138545) in both lines
02:47.42EdeneyeBut I still have this weird number appearing when I call
02:53.47SamotWell, if Nexmo is expecting P-Asserted-Identity from you, you don't have it turned on.
02:54.16Samotfrom_user = unknown <-- did they tell you to set that?
02:54.25SamotBecause that's kind of strange.
03:01.05*** join/#asterisk Janos (~textual@201.204.94.76)
03:13.20EdeneyeIts on their documentation
05:14.07*** join/#asterisk TriJetScud (~TriJetScu@23.172.144.5)
05:35.47*** join/#asterisk Dovid (~dovid@96.56.22.146)
07:12.59jkroonEdeneye, also, E164 format ... so for an SA number 0120210000 I would use CALLERID(num)=+27120210000 ... and similar for PAI (where PAI is used for logging the identity of the party causing the INVITE, and CALLERID(num) the identity of the call origin ... which can be different eg in case of call forwards).
07:25.14*** join/#asterisk Ner0Zer0 (~Ner0Zer0@87.253.63.54)
11:12.55*** join/#asterisk AsteriskRoss (~AsteriskR@37.157.48.2)
11:36.12*** join/#asterisk BakaKuna (~Thunderbi@86-89-65-191.fixed.kpn.net)
12:39.23*** join/#asterisk Edeneye (58a28967@88.162.137.103)
14:26.35*** join/#asterisk pchero (~pchero@211.178.226.108)
14:58.00seanbrightso we're trying to help someone who wants to spoof caller id? nice
15:00.10SamotI did it just to shut them up. Didn't realize it was all a "spoof callerid" thing.
15:00.59igcewieling*shrug* It is the fault of the provider if they allow that.
15:01.10SamotWhat?
15:01.17SamotThat's a bunch of BS.
15:03.46seanbrightwasn't trying to start a thing
15:03.53SamotIt's not a thing.
15:04.31igcewielingSamot and I disagree on almost everything.
15:06.08seanbrightroger that
15:06.19SamotWell, how often are you offering up SIP trunks to those that could be just doing termination with you?
15:07.34SamotI have an emergency alerts provider that I do termination for.
15:07.43SamotMultiple counties in multiple states.
15:08.13SamotPrisons, military bases/contractors, schools, etc..
15:08.18igcewielingThey use a TN assigned to them by us/the carrier.
15:08.30SamotThat's not how that works man.
15:08.36SamotNot for termination only.
15:10.35igcewieling*shrug* If you want to make an outgoing call on our network, the CID will be a number assigned to your account or we override the CID with a default CID on the account.
15:10.57SamotYes, that is very traditional.
15:11.01SamotVery POTS/PRI.
15:12.27SamotThat is something that I do for FXS equivalent services that end up terminating to ATA/FXS gateways.
15:13.21SamotHowever, when I have a SIP trunk end user that just wants a SIP trunk to make calls over. That is what they get.
15:13.59igcewieling*nod* We have 1 of that kind of customer.
15:14.12SamotNow because of the new E9-1-1 laws I have to have at least one registered location. Since I can do off-net 911, they can use their own or I can provide them with an emergency number.
15:15.37igcewieling99% of our VoIP customers are 1) sip to an on-site adtran with analog or pri handoff to the customer or 2) SIP phone connected to a FreePBX box or our hosted server.
15:28.54SamotHonestly though, the TRACED Act has mad a big impact on this already.
15:29.59SamotMade*
15:33.02SamotThe removal of a lot of the rules that limited what they FCC/FTC could do has had a big impact.
15:33.20*** join/#asterisk rpifan (~rpifan@p200300d2672b5d001bf22292c5ddd2f4.dip0.t-ipconnect.de)
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