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| 00:41.08 | JGalt | Where can I find definitively which carrier a us phone number is ported to? anyone able/willing to do SS7 DIPs |
| 00:46.35 | aoeui | https://freecarrierlookup.com/ ? |
| 00:50.24 | JGalt | @aoeui do they actually DIP? |
| 00:51.31 | JGalt | I tried them and they say qwest/centurylink, but qwest/centurylink says the number is "not in their switch. |
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| 07:44.58 | JGalt | a bit of research indicates that the correct SS7 point code would be really helpful. any idea where I can find a corresponding point code given the OCN, CLLI, LRN, erc.? from that I can get carrier contact info similar to an ip whois qwery. |
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| 12:18.59 | Samot | JGalt: What are you trying to do? |
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| 14:07.12 | LunaLovegood | I have some Grandstream phones that periodically send small data packets to port 5060 on my server. It's UDP with usually 28 bytes of binary data as payload, not plain-text SIP. Anyone knows what's up with that? |
| 14:07.42 | LunaLovegood | It's the server where they REGISTER their SIP account and all, so it's not hacking attempts or anything. |
| 14:08.12 | file | NAT keepalive? |
| 14:09.05 | Samot | Have you run a debug or trace? |
| 14:10.17 | LunaLovegood | The phone itself doesn't provide much debugging info, (and there's nothing wrong with the server). It's probably NAT keepalives like file said. Dunno why they wouldn't use OPTION or something else valid though. |
| 14:10.40 | file | because processing a SIP message incurs a processing penalty |
| 14:11.18 | file | so doing some like that can have the remote side do minimal processing, essentially drop it immediately |
| 14:11.34 | Samot | I meant on the server. |
| 14:11.47 | Samot | Have you looked to see what these packets being sent are? |
| 14:12.02 | LunaLovegood | That makes sense. |
| 14:12.23 | LunaLovegood | The packet themselves are like: 4d0d 0001 0000 692c 291b 58af da06 4960 763e bbc6 a2c9 |
| 14:12.39 | LunaLovegood | Dunno why it's not just sending empty packets |
| 14:12.50 | LunaLovegood | (that was the UDP payload) |
| 14:13.13 | LunaLovegood | the whole IP packet is 48 bytes in this case |
| 14:13.16 | Samot | OK well turn on the channel driver debug in Asterisk... |
| 14:13.21 | Samot | See if it's hitting Asterisk.. |
| 14:13.32 | Samot | Or just do a tcpdump to see what those packets are... |
| 14:13.51 | LunaLovegood | Yeah, it's with tcpdump that I saw them. |
| 14:13.58 | Samot | Verbose it. |
| 14:14.04 | Samot | So you can see the actual packet data. |
| 14:14.08 | LunaLovegood | I was troubleshooting something else at the time. |
| 14:14.34 | LunaLovegood | That what I did. It's the non-ascii stuff I posted earlier |
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| 14:15.03 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:15.05 | LunaLovegood | Another example: d665 0001 0000 e706 bdc8 5068 71c3 db91 6f51 d829 1e8c |
| 14:15.39 | Samot | That means nothing. |
| 14:15.41 | LunaLovegood | SIP is supposed to be plain-text, so I don't know what info they could be sending. |
| 14:16.13 | Samot | tcpdump -nttvvvv -s 0 -i eth0 udp |
| 14:16.17 | LunaLovegood | I don't really care since it's sent over a VPN and those packets don't reach farther than my server. |
| 14:16.24 | LunaLovegood | I was just curious. |
| 14:16.27 | Samot | change the eth0 to whatever your interface is.. |
| 14:16.33 | Samot | Is should have more details... |
| 14:17.36 | LunaLovegood | I called it with tcpdump -AXvvvvnn -ixlan 'udp port 5060' |
| 14:18.04 | LunaLovegood | tcpdump can't really decode anything from those packets either, I just get the hexdump |
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| 15:04.03 | JGalt | @Samot Basicly trying to obtain a specific phone number and getting run around in circles by the sales droids who can't seem to figure out that the number really *IS* theirs. I find it really helps in theese cases to start on the engineering side who then calls the sales droid and gives them proper direction. |
| 15:04.38 | Samot | OK but here is the problem with that you can run into. |
| 15:04.58 | Samot | Just because a number belongs to Bandwidth doesn't mean it's freely there for them to give up and port out. |
| 15:05.39 | Samot | A number is going to show the SPID/OCN it is assigned to. |
| 15:06.30 | Samot | That carrier could be providing that DID to a wholesaler/partner.... |
| 15:06.42 | JGalt | interesting..... why not? in this case sales claims that it's "not on their switch" |
| 15:07.04 | Samot | So you looked up the DID can got its SPID/OCN? |
| 15:07.12 | Samot | s/can/and/ |
| 15:07.50 | JGalt | so how does one find said wholesale partner? |
| 15:08.16 | Samot | Well first, do you own this number? |
| 15:08.18 | JGalt | yes, I have the OCN/SPID..... IT'S THEIRS |
| 15:09.02 | JGalt | the number is presently not in service |
| 15:09.19 | Samot | So you're trying to purchase this number? |
| 15:09.44 | JGalt | need to put it in service so it can be ported |
| 15:09.47 | JGalt | yes |
| 15:10.16 | Samot | OK, so what carrier does this DID show belonging to? |
| 15:10.41 | JGalt | basicly set it up as a remote call forward then port it out |
| 15:11.37 | Samot | The other thing you have to keep in mind is that just because a DID as been assigned to an OCN/SPID doesn't mean said carrier has programmed that number in their switch. |
| 15:11.52 | Samot | So they could very well be telling the truth when they say it's not in their switch. |
| 15:11.57 | JGalt | centurylink, but centurylink says "it's not in our switch" |
| 15:12.11 | Samot | Because it might not be programmed in their switch. |
| 15:12.33 | Samot | Carriers just don't put 200K of DID inventory in their switches. |
| 15:14.32 | Samot | Have you talked to CenturyLink sales and requested a service and that specific DID for it? |
| 15:16.03 | JGalt | so centurylink has the thousand block of interest, and the routing info I pulled gives them as the carrier complete with the OCN/SPID verifying back to them. but "it's not in their switch" so how does one get that fixed? what is the problem just putting it in the switch? |
| 15:18.14 | JGalt | yes and they tell me "it's not in our switch" we can't service it, the number is *NOT* ours, we can't service it..... even though the routing info I have shows differently |
| 15:19.13 | Samot | Where did you lookup the DID at? |
| 15:21.30 | JGalt | looked it up with local calling guide and a few others then someone here helped a bit with a look at the SS7 routing database info. |
| 15:21.47 | JGalt | all agreed |
| 15:22.08 | JGalt | everything points to centurylink |
| 15:24.17 | Samot | Can you give me the DID, I want to just double check. |
| 15:35.50 | Samot | JGalt: You are talking to the right part of CenturyLink, correct? |
| 15:37.28 | Samot | JGalt: I can tell you from experience that when I got routed to CL engineering in the West they had to route me back to CL engineering in the Mid West because they had no clue. |
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| 18:47.00 | Penguin | Will an AudioCodes MP-202 connect the SIP lines via LAN port or only the WAN port? I don't want to use it as a router for the computer network; I only want to use it as an ATA. |
| 18:54.43 | Samot | You use the WAN port. |
| 18:54.56 | Penguin | That's what I'm finding out during some testing. |
| 18:55.06 | Samot | You would only make it a router if you replace your router with it. |
| 18:55.21 | Samot | Otherwise the WAN will get an IP from your router's DHCP server. |
| 18:55.43 | Penguin | I had to connect up to the LAN port to turn on the WAN port remote administration so I can access it once I connect the WAN port to the local network. |
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| 19:13.22 | [TK]D-Fender | That's the standard procedure. Connect local, enable admin on Wan Interface, replug and just manage off WAN |
| 19:15.19 | Penguin | The next part is why TLS doesn't work. UDP works, but TLS gives an SSL error on the asterisk console. And there's no SRTP setting in the ATA. I think maybe it needs a firmware upgrade. |
| 19:16.30 | Penguin | Current Version: 2.6.4_p041_build_9 |
| 19:17.28 | Penguin | Looks like they have version 3.0.3 p070 build 12 on the web site for download. |
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| 20:02.53 | Penguin | Nope, now it works even less. Still no SRTP even though the manual shows it, and TLS never tries to register. |
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| 20:40.49 | Meaw | hello |
| 20:41.04 | Meaw | Need help with originate command through the asterisk cli |
| 20:41.50 | Meaw | trying to place a call through originate command to pstn line, but its not working.. I tried " originate SIP/number " but not working |
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| 21:42.10 | Meaw | anyone here? |
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| 21:58.53 | kharwell | yes |
| 22:01.30 | Meaw | trying to place a call through originate command to pstn line, but its not working.. I tried " originate SIP/number " but not working |
| 22:01.51 | Meaw | the cli return with this : Usage2: originate <tech/data> extension [exten@][context] |
| 22:02.02 | Meaw | which I dont really understand. |
| 22:05.21 | kharwell | sounds like you didn't format the command properly. Here's a handy guide and example: https://wiki.asterisk.org/wiki/display/AST/Creating+and+Manipulating+Channels+from+the+CLI |
| 22:05.27 | kharwell | see section on "originate" |
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| 22:08.25 | Meaw | yes im not asterisk expert, its just i want to test the pstn lines through the cli coz I cannot register to sip because it is internal network |
| 22:08.38 | Meaw | will read the guide thank you |
| 22:09.00 | kharwell | also note similar help can be found from the CLI using the "core show help ...." for that command and others. For instance CLI> core show help channel originate |
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| 22:19.46 | Meaw | kharwell, yes read the guide, i've an embedded asterisk, I use originate not channel originate, but whats is 9999@somecontest? |
| 22:20.25 | Meaw | I tried originate SIP/SIPuser extension 9999@default and I got this result |
| 22:20.34 | Meaw | [Jun 6 18:18:41] NOTICE[14064]: channel.c:2863 __ast_request_and_dial: Unable to request channel SIP/210 |
| 22:25.19 | kharwell | so when you use that command a channel within Asterisk is originated for SIP/SIPuser. In this case SIPuser must be defined somewhere within sip.conf . If that peer (?) has a known way of reaching the device then it will dial it. 9999@somecontest takes you into the dialplan itself to extension 9999 within context "somecontest". Whatever you have there is then processed |
| 22:25.47 | Penguin | I'm tired of ATAs not doing TLS and SRTP. What kind of one or two line ATA should I get that does TLS and SRTP and works with asterisk? |
| 22:31.47 | Penguin | I don't need anything fancy. I have a Cisco that handles TLS just fine, but doesn't seem to do SRTP with asterisk even though it has an encryption setting. |
| 22:32.38 | Penguin | I have an AudioCodes that has settings for TLS, but it doesn't work, and the manual shows it has SRTP, but the UI doesn't have it available. |