| 05:48.36 | *** join/#asterisk infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net) |
| 05:48.36 | *** topic/#asterisk is AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.30.0 (2019/12/23) 16.7.0 (2019/12/23) Standard: 17.1.0 (2019/12/23); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 06:02.10 | rastael | @samot: thanks for replying... the list i have is 10digits like 7703394488 , I tried changing it to exten:7 .. and it adds numbers in the logs.........So i changed it to exten:13.... and looking at plivo logs it works +7703394488 however i cant get a dialtone because the asterisk log shows -SIP 770 Prefix Not Supported) completed, returning 0 .............. it only works if i dial |
| 06:02.10 | rastael | 01101117703394488 with exten => _011.,2,Dial(SIP/${EXTEN:10}@plivo,,tTo) ............. I simply want to do dial 7703394488 .... not sure if there is a way to incorporate the 011 automatically ? |
| 06:03.44 | rastael | when lowering the strip to 7 .. i end up with +12347703394488 for example.. but using the 13 makes the number 10 digit.. but no dialtone .. |
| 06:04.41 | Samot | <PROTECTED> |
| 06:04.55 | Samot | Which would be ${EXTEN:7} |
| 06:05.28 | Samot | That will leave you 7703394488 so I have no idea where the 1234 is coming from. |
| 06:06.24 | Samot | Are you trying to call the US from outside the US/Canada? |
| 06:07.22 | rastael | @samot: am using asterisk + goautodial .. calling usa to usa from plivo.. this is the current extension that am using.. |
| 06:07.26 | rastael | exten => _011.,1,AGI(agi://127.0.0.1:4577/call_log) |
| 06:07.26 | rastael | exten => _011.,2,Dial(SIP/${EXTEN:13}@plivo,,tTo) |
| 06:07.26 | rastael | exten => _011.,3,Hangup |
| 06:07.57 | rastael | i simply want to dial any USA 10digit number, without inserting 011 or a 1 at the beginning .. |
| 06:08.17 | [TK]D-Fender | Get rid of the 011 then |
| 06:08.19 | Samot | Why do you need to match against 011 first? |
| 06:08.22 | [TK]D-Fender | You put it there |
| 06:08.29 | Samot | If you're calling from the US to the US it's not needed. |
| 06:08.36 | rastael | @samot: i was following the plivo asterisk intergration instructions. |
| 06:08.38 | Samot | You can send calls with 11 or 10 digits. |
| 06:08.42 | [TK]D-Fender | exten => _011., <- that's not a 10-digit pattern |
| 06:09.01 | [TK]D-Fender | And an unlimited length, but 4 minumium with crap in front |
| 06:09.04 | [TK]D-Fender | Stop doing that |
| 06:09.13 | [TK]D-Fender | If you want 10 digits... MAKE it 10 digits |
| 06:09.17 | rastael | ok |
| 06:09.31 | [TK]D-Fender | having a "." in there mean "1 or MORE after" which is unlimited |
| 06:09.32 | Samot | rastael: Is plivo requiring a prefix? |
| 06:09.48 | Samot | Or a tech code or what every they are calling it? |
| 06:09.56 | Samot | ever* |
| 06:10.14 | rastael | _NXXNXXXXXX would be it then? |
| 06:10.34 | Samot | First, who is dialing? |
| 06:10.45 | Samot | Because you're trying to match what an endpoint is sending you, correct? |
| 06:10.49 | rastael | i have goautodial.. am dialing from agent console.. |
| 06:10.50 | Samot | So they can make an outbound call? |
| 06:10.55 | [TK]D-Fender | rastael, that is a 10-digit pattern at least |
| 06:10.58 | Samot | So who is adding the prefix? |
| 06:11.03 | [TK]D-Fender | And the based for NANPA dialing |
| 06:11.15 | rastael | i am.. i dont want a prefix.. i want any 10digit to dial outbound |
| 06:11.34 | Samot | So why are you adding it? |
| 06:13.55 | Samot | Also, just to verify. This is a legal and valid use case for a predictive dialer? |
| 06:14.01 | rastael | honestly because after adding the carrier on goautodial.. it autocreated it.. and i was trying to figure out a way not to break it trouble shooting different combinations. |
| 06:15.23 | rastael | @samot: yes i have a real estate company and am tired of paying mojosells for a tripleline dialer.. i am intergrating goautodial + vtiger and some mautic to have everything inhouse |
| 06:15.33 | rastael | @samot: yes i do dnc scrub |
| 06:23.29 | Samot | Well this sounds like a goautodial issue if they are inserting the prefix |
| 06:27.49 | Samot | rastael: What version of Asterisk are you running? |
| 06:33.59 | rastael | @samot: 13.x .. ok i think i see something... |
| 06:34.22 | rastael | @samot: when using this dialplan exten => _011.,1,AGI(agi://127.0.0.1:4577/call_log) |
| 06:34.23 | rastael | exten => _011.,2,Dial(SIP/${EXTEN:10}@plivo,,tTo) |
| 06:34.23 | rastael | exten => _011.,3,Hangup |
| 06:34.46 | rastael | and calling 0010011<number> it works... and plivo log shows +1<number> |
| 06:34.59 | rastael | when i change to this dialplan |
| 06:35.00 | rastael | exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) |
| 06:35.01 | rastael | exten => _1NXXNXXXXXX.2,Dial(SIP/${EXTEN}@plivo,,tTo) |
| 06:35.01 | rastael | exten => _1NXXNXXXXXX,3,Hangup |
| 06:35.46 | rastael | well without the one.. it makes plivo show +<number> ... |
| 06:36.24 | rastael | the call works only when plivo logs show +1<number> ... not +<number> ...... how can i add the 1 ... into this dial plan |
| 06:36.25 | rastael | exten => _NXXNXXXXXX.2,Dial(SIP/${EXTEN}@plivo,,tTo) |
| 06:38.43 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 06:39.38 | Samot | You put a 1 in front of ${EXTEN} |
| 06:39.46 | Samot | 1${EXTEN} |
| 06:56.29 | rastael | okk we are in business now.. i got it working |
| 06:56.30 | rastael | exten => _011.,1,AGI(agi://127.0.0.1:4577/call_log) |
| 06:56.31 | rastael | exten => _011.,2,Dial(SIP/1${EXTEN:13}@plivo,,tTo) |
| 06:56.31 | rastael | exten => _011.,3,Hangup |
| 06:56.35 | rastael | did the magic... thank you samot |
| 07:12.34 | *** join/#asterisk pchero_work (~pchero@87.213.247.82) |
| 07:23.11 | *** join/#asterisk Asterisco (~catan@host170-83-dynamic.36-79-r.retail.telecomitalia.it) |
| 07:23.21 | Asterisco | Good morning |
| 07:24.02 | Asterisco | i've a problem when i recive a call in a voip number... asterisk try answer with another number ?!?!?!? |
| 07:24.21 | [TK]D-Fender | fix your description and show us the problem |
| 07:24.23 | Asterisco | <PROTECTED> |
| 07:24.25 | *** join/#asterisk tsal (~tsal@i59F4A24D.versanet.de) |
| 07:24.35 | Asterisco | i've two voip number |
| 07:24.53 | Asterisco | the first is 0955183828 and work fine |
| 07:25.03 | Asterisco | the second is 0952830382 |
| 07:25.38 | Asterisco | but when i recive a call at this number.... i obtain this log: -- Executing [0952830382@incoming:1] Answer("SIP/0955183828-00000000", "") in new stack |
| 07:25.38 | Asterisco | <PROTECTED> |
| 07:26.19 | Asterisco | is strange becouse in sip.conf i've set all right |
| 07:26.30 | Asterisco | and in extension is the same |
| 07:26.56 | Asterisco | exten => 0952830382,1,Answer() |
| 07:26.57 | Asterisco | exten => 0952830382,n,Dial(SIP/31,60) ; Squilla il cordless di via Morandi |
| 07:26.57 | Asterisco | exten => 0952830382,n,Hangup() |
| 07:27.17 | [TK]D-Fender | because that is what your provider is sending |
| 07:27.31 | Asterisco | ??? |
| 07:27.43 | [TK]D-Fender | What is not clear about that simple statement? |
| 07:27.56 | Asterisco | the provide send only a number for ip address? |
| 07:28.08 | [TK]D-Fender | Executing [095283038 <- THIS ..... is what your provider is SENDING |
| 07:28.29 | [TK]D-Fender | <PROTECTED> |
| 07:28.35 | Asterisco | one moment.... |
| 07:28.42 | [TK]D-Fender | (chopped digit by accident) |
| 07:28.44 | Asterisco | Executing [0952830382@incoming:1] Answer("SIP/0955183828-00000000", "") in new stack |
| 07:29.00 | Asterisco | see SIP/0955183828 |
| 07:29.15 | Asterisco | is not the right number |
| 07:29.17 | [TK]D-Fender | You can see which PEER was matched and what NUMBER the provider is sending you in thier "dial" |
| 07:29.30 | [TK]D-Fender | it SUCCESSFULLY matched the FIRST peeer |
| 07:29.52 | [TK]D-Fender | you obviously aren't AUTHING calls differently between them on incoming |
| 07:29.58 | [TK]D-Fender | otherwise it wouldn't have matched |
| 07:30.44 | Asterisco | 0952830382/0952830382 83.211.227.21 Yes Yes 5061 UNREACHABLE |
| 07:30.55 | Asterisco | 0955183828/0955183828 83.211.227.21 Yes Yes 5060 OK (34 ms) |
| 07:32.00 | Asterisco | ok now is the same port 5060 |
| 07:32.16 | Asterisco | and both are reachable |
| 07:32.38 | [TK]D-Fender | What part of "the first one SUCCESSFULLY matches" is unclear? This has NOTHING to do with "unreachable" status |
| 07:32.50 | [TK]D-Fender | that only stops * for caling OUT to that peer |
| 07:32.58 | [TK]D-Fender | changes nothing for matching incoming calls |
| 07:33.26 | Asterisco | TK i don't understand |
| 07:33.37 | [TK]D-Fender | This ifirs t one MATCHES |
| 07:33.41 | [TK]D-Fender | What is there to not understand? |
| 07:33.45 | [TK]D-Fender | A peer matches by IP |
| 07:33.51 | [TK]D-Fender | the IP of your FIRST peer matches |
| 07:33.54 | Asterisco | is normal? |
| 07:34.02 | [TK]D-Fender | * won't give a shit or evenm LOOK at the second entry |
| 07:34.16 | Asterisco | bt is normal? |
| 07:34.19 | Asterisco | but |
| 07:34.22 | [TK]D-Fender | FIRST come = GAME OVER |
| 07:34.27 | [TK]D-Fender | no "buit" |
| 07:34.32 | [TK]D-Fender | There is no "but" |
| 07:34.41 | [TK]D-Fender | It doesn't look at "number" |
| 07:34.55 | [TK]D-Fender | "number" means nothing |
| 07:35.00 | [TK]D-Fender | peer = match by IP |
| 07:35.14 | Asterisco | ...ok |
| 07:35.47 | Asterisco | so is impossible recive calls in the second number? |
| 07:35.53 | Asterisco | phone number? |
| 07:35.57 | [TK]D-Fender | stop saying "number" |
| 07:36.04 | [TK]D-Fender | that doesn't mean anything |
| 07:36.12 | [TK]D-Fender | your PEER NAME is a number |
| 07:36.14 | Asterisco | so is impossible recive calls in the second phone number? |
| 07:36.27 | [TK]D-Fender | they are DIALING an extenion NUMBER into your dialplan |
| 07:36.39 | [TK]D-Fender | Asterisco> so is impossible recive calls in the second phone number? <- You are GETTING THE CALL |
| 07:36.42 | [TK]D-Fender | And stop saying NUMBER |
| 07:36.44 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 07:36.46 | [TK]D-Fender | EVERYTJHING is a number in there |
| 07:37.01 | Asterisco | TK i recive a call MUTE |
| 07:37.10 | Asterisco | this is the problem |
| 07:37.24 | [TK]D-Fender | Why are you only saying this NOW? |
| 07:37.30 | [TK]D-Fender | YOu've been talking about a stupid NUMBER |
| 07:37.40 | Asterisco | ok is a my problem! |
| 07:37.48 | Asterisco | my english is not good |
| 07:37.52 | [TK]D-Fender | for the language |
| 07:37.59 | [TK]D-Fender | you aren't even talking about the right TOPIC |
| 07:38.01 | Asterisco | now i tell you that i recive a mute call |
| 07:38.14 | [TK]D-Fender | Look at the call |
| 07:38.19 | Asterisco | or sometime the call hungup immediatly |
| 07:38.39 | Asterisco | how can i look at the call? |
| 07:39.03 | [TK]D-Fender | same as aloways. SIP DEBUG |
| 07:39.09 | Asterisco | ok |
| 07:39.34 | Asterisco | No such command 'sip debug' (type 'core show help sip debug' for other possible commands) |
| 07:40.53 | Asterisco | sip set debug on |
| 07:40.58 | Asterisco | i try :) |
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