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| 04:30.27 | _marshy | anyone using flowroute know how to remove the "1" on outbound calls? e.g. CNAM is set to 2223331234, CID shows up as 12223331234 |
| 04:30.29 | _marshy | ? |
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| 04:34.09 | Samot | Well that is just plain ole digit manipulation. |
| 04:35.05 | Samot | You're also confusing CNAM with CID |
| 04:35.11 | Samot | They are two separate things. |
| 04:35.23 | Samot | CID is the actual number, CNAM is the name associated with that number. |
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| 04:35.53 | Samot | You have to send calls to Flowroute in 1NXXNXXXXXX format |
| 04:36.13 | Samot | You can't control how the other side presents the CallerID Number. |
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| 04:37.20 | Samot | You can try: "Name" <number> format in your ourbound CallerID |
| 04:37.54 | Samot | Set(CALLERID(full)="Name" <number>) |
| 04:39.04 | Samot | But if the other side ignores that and pulls from the TO header or manipulates it to be 1NXXNXXXXXX, you can't control it. |
| 04:39.36 | Samot | Sorry, FROM header. |
| 04:40.11 | _marshy | the number needs to be in <> brackets? |
| 04:40.39 | Samot | Full, proper CallerID with name is "Name" <number> |
| 04:41.19 | ChannelZ | They don't give a crap about the name |
| 04:41.25 | Samot | Also make sure your trunk has sendrpid=pai |
| 04:41.33 | ChannelZ | unless possibly flowroute<->flowroute SIP native calls |
| 04:41.36 | ChannelZ | if they even do that |
| 04:42.53 | lvlinux | Actually sometimes Comcast Voice customers will get the name passed through, when called from a different carrier. |
| 04:43.23 | Samot | The terminating carrier will touch the callerid |
| 04:43.34 | Samot | To send it downstream. |
| 04:43.43 | Samot | They very well could be manipulating this. |
| 04:43.51 | _marshy | hmm |
| 04:43.58 | Samot | I have a carrier that sends callerid in E.164 |
| 04:44.04 | Samot | Doesn't matter how you set it.. |
| 04:44.10 | Samot | They present it to me in E.164 |
| 04:44.17 | Samot | So I have to strip the +1 |
| 04:44.21 | Samot | If I want to present 10 digits |
| 04:44.51 | _marshy | food for though |
| 04:44.52 | _marshy | t |
| 04:44.57 | Samot | Just like if my end users send me a call... |
| 04:44.57 | ChannelZ | oh that reminds me |
| 04:45.05 | Samot | And RPID or PAI aren't set.. |
| 04:45.15 | Samot | I set them before sending them out |
| 04:45.30 | ChannelZ | What was that one protocol that was like DNS for phone numbers, so you could look up a PSTN number and possibly get a direct SIP route for it |
| 04:45.38 | ChannelZ | I wish that'd take off |
| 04:45.39 | Samot | ENUM |
| 04:45.42 | ChannelZ | yeah that's it |
| 04:46.15 | Samot | It wasn't "like DNS" it is DNS |
| 04:46.16 | ChannelZ | I guess that's techincally part of E.164 |
| 04:46.25 | Samot | Well it uses E.164 |
| 04:46.33 | Samot | Because E.164 is universal |
| 04:47.22 | Samot | E.164 for a UK number is the same format, albeit more digits, as a E.164 US number. |
| 04:48.34 | Samot | ENUM had no centralized routing |
| 04:48.50 | Samot | That was one of its flaws. |
| 04:50.18 | Samot | My ENUM records were only good for routing people who were looking up calls in my ENUM servers. |
| 04:51.37 | ChannelZ | well exactly, that it's reverse DNS means it's gotta terminate at one standard domain that someone has to be the master of for it to be useful |
| 04:52.03 | Samot | Well.. |
| 04:52.16 | Samot | ENUM requires NAPTR and SRV records as well.. |
| 04:52.21 | ChannelZ | I'm just browsing and someone's come up with a blockchain type method to decentralize it |
| 04:53.04 | ChannelZ | but that's probably even more esoteric for it to catch on |
| 04:53.22 | Samot | ENUM was never going to pass mustard. |
| 04:53.37 | Samot | You're talking about a collective of global carriers. |
| 04:54.02 | Samot | And an even smaller collective that regulates how those carriers interact.. |
| 04:54.36 | Samot | Someone controls how DIDs are assigned in the US |
| 04:54.42 | Samot | And someone else in the UK |
| 04:54.59 | ChannelZ | Well yeah, there's a lot of self-interest involved, on all sides :) |
| 04:55.02 | Samot | They also control the centralized routing for those. |
| 04:55.31 | Samot | They are the "root servers" of the PSTN |
| 04:55.48 | Samot | If you want to equivocate DNS terms with this. |
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| 04:57.55 | Samot | I've worked at CLECs and the amount of money we spent for our infrastructure was insane... |
| 04:57.57 | ChannelZ | Well the carriers don't matter particularly. It's really just a DNS problem of having someone in control of a common domain that all the mappings live under. |
| 04:58.28 | Samot | And that's a CLEC, who has to "rent" infrastructure from the ILEC... |
| 04:58.29 | ChannelZ | It's not in a carrier's interest to do it because you're basically using it to bypass them |
| 04:58.41 | Samot | So what an ILEC puts into it, probably 10x more. |
| 04:58.48 | Samot | OK. |
| 04:58.58 | Samot | So then where do I get my number? |
| 04:59.06 | Samot | Now I'm just free to make it up? |
| 04:59.08 | Samot | No rules? |
| 04:59.10 | Samot | Awesome. |
| 04:59.16 | Samot | It's a public utility. |
| 04:59.17 | ChannelZ | The same place you do now, if you want a PSTN presense |
| 04:59.22 | ChannelZ | *presence |
| 05:01.51 | ChannelZ | I mean I see what your saying and don't disagree.. it would require policing, which would require carrier interference, which has no interest in helping people bypass their own services :D |
| 05:02.06 | Samot | Not even carrier interference.. |
| 05:02.18 | Samot | The carriers *answer* to higher bodies. |
| 05:02.45 | Samot | I can see the carriers lobbying against it for valid reasons.. |
| 05:02.54 | ChannelZ | of course |
| 05:03.08 | Samot | But the control bodies that handle how routing/assignment/etc is done would have the biggest say |
| 05:03.18 | Samot | Since *they* would be the ones regulating it |
| 05:04.25 | Samot | They've already create a "VoIP" class |
| 05:04.30 | Samot | They've already created a "VoIP" class |
| 05:05.31 | Samot | So carrier class/type denoting what the numbers are allocated for (originally) are tagged to DIDs. |
| 05:06.14 | Samot | So a pure wireless carrier has their numbers marked "Wireless" |
| 05:06.27 | Samot | A normal carrier "Landline" |
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| 05:06.35 | Samot | And a pure IP/VoIP as "VoIP" |
| 05:07.06 | Samot | That's why a lot of "voip" numbers come back as "Landline" because that carrier that owns it, is that type of carrier. |
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| 05:08.15 | Samot | So far Flowroute is the only registered "VoIP Only" CLEC in the US. |
| 05:08.22 | Samot | That I know of. |
| 05:08.42 | Samot | And technically only in a region of Nevada. |
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| 08:35.17 | v0lZy | hi everyone |
| 08:36.15 | v0lZy | Have a quick question ... I've been talking to some VoIP providers about DID, and some say one has to register the internal numbers with them, while some say it is not necessery |
| 08:39.36 | v0lZy | what is the general practice, if say you have a single registered phone number, and 10 internal numbers ... I'm assuming an ITSP can easily match '_XXX XXX XXX.' and thereby the client should have complete freedom in having as many internal extensions as they want. |
| 08:40.27 | v0lZy | or would they have to register XXX XXX X01, XXX XXX X02, XXX XXX X03, .. XXX XXX X10 with the ITSP? |
| 08:41.26 | v0lZy | I would expect DID being restricted on the number of channels, but otherwise not limit the internal extensions of the client |
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| 08:42.35 | v0lZy | (I'm trying to move from my current ITSP where we only have 1 registered number and can spawn internal extensions at will, to a different provider that allows for the same thing... now some say they allow that, some say they dont) |
| 08:42.47 | v0lZy | (but maybe im using the wrong lingo with the ones that say they dont, I dont know) |
| 08:45.37 | bluez_ | hmm is it possible to dial an extension 3 times, connect back the first one that connects, but let the second one ring for upto 2 seconds? |
| 08:46.06 | bluez_ | i know i can dial multiple extensions and connect back the first one and hangup the rest with dial() |
| 08:47.01 | v0lZy | bluez_: not sure, but maybe you could create a conference call, then dial everyone you want and if they pick up bridged them into the conference |
| 08:47.14 | v0lZy | bluez_: and if they dont pick up after 2 seconds, stop the ringing. |
| 08:48.14 | bluez_ | hmm interesting, then i could control what to do with each call depending on pickup, ring etc |
| 08:48.21 | bluez_ | can i do a co |
| 08:48.27 | bluez_ | a condition on ring? |
| 08:50.34 | bluez_ | actually all i need to do is dial extension 500 3 times, first one that connects bridge it back to the caller, the first one to ring of the two left, let it ring twice and then hangup those 2 |
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| 08:58.44 | v0lZy | what is the use case for this thought? |
| 08:58.47 | v0lZy | though* |
| 09:00.40 | v0lZy | bluez_: I think the way it would work with conference would be that you bridge the caller into the conference room, then play them some music or whatever, while you call other 3 extensions .. the first one to pick up is bridged into the the conference so they can talk with the caller |
| 09:01.37 | v0lZy | and the remaining two you can then timeout, or whatever... look at how many participants are in the conference call and if its 2, simply not bridge them in or sometihng |
| 09:01.49 | bluez_ | basically there's 3 phones. when a call comes in, all 3 ring. once is sometimes put to voicemail, but we want the second one to ring once so we know a call came in (and was directed to voicemail) |
| 09:02.20 | bluez_ | so the one ring on exten2 is more if a signal on that phone that we had a call on line 1 and it's gone to vm |
| 09:02.58 | v0lZy | on my phone, if I have voice mail i get a blinking light. |
| 09:03.04 | v0lZy | then I check voicemail |
| 09:03.08 | v0lZy | you dont have that? |
| 09:03.23 | bluez_ | yeah it's basically because eten1 dials another platform which may or may not be in vm |
| 09:03.38 | v0lZy | well |
| 09:03.55 | v0lZy | I have here on the phones also the option to set sometihng direct to voicemail |
| 09:04.02 | v0lZy | and what you could do is pass through dialing o rsometihng like that |
| 09:04.06 | bluez_ | let me explain again maybe. Let's say we have a phone on the front desk which may locally be sent to vm or not. |
| 09:04.52 | v0lZy | you ring the hunt grup, and if none of the hunt group answer within X rings, send call to special extension, then from that special extension to voicemail |
| 09:05.28 | v0lZy | so call comes in, you ring the group, noone picks up, so you ring the designated phone, which either answers within 2 sec, or you put the call in voicemail |
| 09:06.13 | v0lZy | that way, you cascade it and if you have unanswered call on the designated phone, you know something went to voicemail |
| 09:06.37 | v0lZy | OK |
| 09:06.42 | bluez_ | ok but we can't direct it to voicemail from asterisk |
| 09:07.01 | bluez_ | ok let's assume line 1 is some other asterisk which may or may not be sent to vm right now |
| 09:07.28 | v0lZy | ok ... line1 is the front desk phone which is either set to VM or not |
| 09:07.30 | bluez_ | when line 1 is not on voicemail: we dial(line1&line2&line3), first one to pickup connects back and the rest hangup - normal case |
| 09:07.44 | bluez_ | when line 1 is on voicemail. we know it will immediently answer |
| 09:07.52 | bluez_ | but we want a signal to line 2 (or 3) that that happened |
| 09:08.23 | bluez_ | so line 2 or 3 will never ring in that case usually (well it's a race they might or might not dependoing on how fast line 1 goes to voicemail, but generally it answeres to quick) |
| 09:09.05 | bluez_ | we're using the imediete answer on line 1 to know it's sent to vm because a human won't answer that quick. it's not ideal but its fine for our uses right now |
| 09:09.20 | v0lZy | In my system, theres two ways to do voicemail... I can set it on the phone, such that after X rings it goes to voicemail, or set it to go to voicemail immidietly etc |
| 09:09.46 | bluez_ | right, so line 1 goes to vm immedietly (and so answers immedietly) we can't set it to x rings |
| 09:10.32 | bluez_ | line 1 isn't a normal phone as such, it's another system - assume we only have instant to vm option |
| 09:10.47 | v0lZy | I see |
| 09:11.17 | bluez_ | basically if i could dial 3 extensions and do something based on if they are ringing, answered or hung up i could taloir this how i need to |
| 09:12.11 | bluez_ | eg dial 1,2,3 first to connect bridge to caller, hangup rest if have rang for 1 second at least |
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| 09:12.14 | v0lZy | bluez_: seeing as line1 is either VM or it bridges the call... is this something users manually set? |
| 09:12.44 | v0lZy | your users I mean. |
| 09:12.47 | bluez_ | sometimes but it's semi-automatic |
| 09:13.31 | bluez_ | assume we can't change config based on if line 1 is going to vm or not, all we have is our guess based on it answering before line 2 and 3 |
| 09:14.00 | v0lZy | So safe to assume that as long as you are calling line1, it will either ring (and thats where you are 'stealing the call' with other two lines) or immidiately go to voicemail, in which case you want to know that that happened? |
| 09:14.02 | bluez_ | the conference idea could work, if i can make decisions based on call 1,2,3's state |
| 09:14.11 | bluez_ | yes |
| 09:14.27 | bluez_ | i want line 2 || 3 to ring at least 1 always |
| 09:14.37 | bluez_ | even if line 1 was picked up before that first ring |
| 09:14.49 | bluez_ | ring once or ring for 1 second etc - whatever is possible |
| 09:15.07 | v0lZy | bluez_: what if you simply threw line1 out of your hunt grup, and for each call, you would call line2+line3, and if they dont answer, then call line1? |
| 09:15.46 | bluez_ | because when not in vm we want 1,2,3 to all ring at the same time |
| 09:16.04 | bluez_ | i know it sounds a bit rediculous lol but i'm simplifying this a bit in terms of what's on the end of line 1 |
| 09:17.00 | v0lZy | I fail to see how this would logically work; you have non deterministic behavior for line1, which you can't determine prior to taking an action |
| 09:17.50 | v0lZy | whenever you ring 1,2,3, you might or might not get answered by 1 (natural pickup or immidiate pickup and bridge to voicemail) |
| 09:18.18 | bluez_ | ok so... |
| 09:18.37 | v0lZy | doesnt seem you can change that with the conference call suggested above either, since the danger here is that line1 flips to voicemail and you dont know about it.. so it will pickup anything as soon as it rings |
| 09:18.47 | v0lZy | in that sense, its greedy |
| 09:18.52 | bluez_ | not in vm: dial 1,2,3 - all ring at least once because a human can't pick up line 1 before 2 and 3 ring |
| 09:19.06 | bluez_ | in vm: dial 1,2,3 - 1 picks up right away and 2,3 hangup before they ring |
| 09:19.24 | v0lZy | yeah, thats clear. |
| 09:20.28 | bluez_ | i'm using lua, i was hoping i could do things based on call status |
| 09:20.33 | bluez_ | (exentisons.lua) |
| 09:20.58 | v0lZy | Dont know, never used lua myself, just extensions.conf |
| 09:21.13 | v0lZy | you could maybe do something else |
| 09:21.34 | v0lZy | Asterisk I think can do triggers... like when there is a file in some folder, it would do something |
| 09:22.05 | bluez_ | oh interesting |
| 09:22.13 | bluez_ | it can't do triggers on call status? |
| 09:22.30 | v0lZy | I dont know, been a while since i altered any dial plans :D |
| 09:23.04 | v0lZy | but with the trigger stuff you could potentially create a file for every call that comes in, and depending on the location you created it at, you could then trigger a certain action |
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| 09:24.56 | bluez_ | yeah |
| 09:25.36 | v0lZy | so you could in theory create a file in a directory that triggers a call to whatever extension you want. |
| 09:26.29 | v0lZy | but again, I dont see that solving your problem |
| 09:27.04 | bluez_ | hmm maybe i should modify the source of dial() |
| 09:27.41 | bluez_ | i would need to delay the hangup of the other channels i guess hmm |
| 09:28.16 | bluez_ | or... |
| 09:29.19 | v0lZy | well |
| 09:29.30 | v0lZy | you could add an extension that delays line1 |
| 09:29.32 | v0lZy | like |
| 09:29.33 | bluez_ | dial(2) with timeout 1 second, then dial(1 & 2 & 3), this will force 2 to ring once and then dial all 3 |
| 09:29.50 | v0lZy | create line1a that after 1 second forwards to line1 |
| 09:30.01 | v0lZy | then in your hunt group ring line1a,line2,lin3 |
| 09:30.02 | v0lZy | line3 |
| 09:30.03 | bluez_ | even better way to do it |
| 09:30.14 | v0lZy | that way you are giving a 1 second advantage to line2 and line3 |
| 09:31.43 | bluez_ | this should work |
| 09:36.36 | bluez_ | is it possible to do a condition on the far end ringing? |
| 09:40.05 | v0lZy | erm... what do you mean? |
| 09:40.53 | bluez_ | so dial (1) , onRing(dosomething) |
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| 09:46.33 | v0lZy | whoops, lost power |
| 09:46.46 | v0lZy | didnt see your last reply bluez_ |
| 09:47.00 | bluez_ | "so dial (1) , onRing(dosomething)" |
| 09:50.36 | v0lZy | mm |
| 09:50.45 | bluez_ | seems we have app_waitRing |
| 09:53.38 | v0lZy | never tried it. |
| 10:15.48 | bluez_ | hmm doesn't work, i think Dial() for sip calls has to return before the next line runs |
| 10:16.45 | v0lZy | probably |
| 10:16.59 | v0lZy | it would be weird if it was non blocking. |
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| 12:08.57 | wasanzy | hello |
| 12:09.22 | wasanzy | when I do OBD call to a phone number, the phone number is not logged in my CDR log. |
| 12:09.29 | wasanzy | anyway I can fix this? |
| 12:12.29 | Samot | OBD? |
| 12:12.34 | Samot | Outbound? |
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| 12:23.50 | *** mode/#asterisk [+o gtjoseph] by ChanServ |
| 12:31.40 | wasanzy | Samot: yes Outbound |
| 12:32.25 | Samot | What field is not populating? |
| 12:32.40 | wasanzy | I have fixed that |
| 12:33.08 | wasanzy | now my problem is when outbound call is made and the person did not pick the call, nothing is logged in the cdr. |
| 12:33.31 | wasanzy | but I have this there disposition |
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| 13:13.26 | bounceman | Hi, can anyone aid me in this. res_musiconhold.c:1438 local_ast_moh_start: Unknown format 'sln' -- defaulting to SLIN |
| 13:13.32 | bounceman | Asterisk dont know sln? I thought it did |
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| 14:48.05 | eric_hill | Has anyone here experienced the joy of integrating Asterisk with DealerTrack? |
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| 14:48.37 | Samot | Nope |
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| 15:01.55 | voipmonk | sounds like fun |
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| 15:20.58 | voipmonk | what are you attempting to do , eric_hill ? |
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| 16:00.07 | eric_hill | voipmonk, a local car dealer is having issues with their current system and asked me to replace the phone system. I'm looking at how that software talks to phone systems and its... different. |
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| 16:00.24 | voipmonk | ok |
| 16:00.28 | voipmonk | explain? |
| 16:00.45 | Dovid | Alsow hat is DealerTrack? |
| 16:01.36 | eric_hill | The DealerTrack software hooks into the call process. You have to send inbound calls to it, and it forwards them back to you to deliver to a phone. All so it can log the call to the caller I. |
| 16:01.47 | eric_hill | caller ID |
| 16:02.04 | voipmonk | what phones are they using? |
| 16:02.05 | eric_hill | The problem is that it is reporting that it *doesn't* support any codecs. |
| 16:02.39 | eric_hill | Polycom and Snom I believe. Haven't been onsite yet. |
| 16:02.43 | voipmonk | make/model of phones, what kind of lines are coming into the building? are they using sip? |
| 16:02.45 | voipmonk | ahh ok |
| 16:02.56 | voipmonk | well do a site survey and come back - |
| 16:02.59 | eric_hill | I've got a call into the DT software to get some more technical details. |
| 16:03.00 | Samot | All this for a CallerID? |
| 16:03.47 | eric_hill | Samot, the software pops up on screen and shows all contact with the customer including email, phone calls, sales people, possible deals, etc. It's spooky how pushy they can be. |
| 16:03.54 | Samot | What is the actual issue? |
| 16:04.13 | eric_hill | The report is "sometimes calls don't go through". |
| 16:04.25 | Samot | Go through where? |
| 16:04.29 | Samot | Are they making it back to the PBX? |
| 16:04.54 | eric_hill | Again, I haven't been onsite yet. I was curious if any of you had touched that integration before I went digging for fools gold. |
| 16:05.10 | Samot | You don't have remote access to the PBX? |
| 16:05.26 | eric_hill | PSTN SIP -> asterisk -> DealerTrack -> asterisk -> SIP phone |
| 16:05.32 | Samot | You don't have remote access to the PBX? |
| 16:05.34 | eric_hill | Samot, not yet. That's coming today. |
| 16:08.38 | Dovid | I would go on site first and work your way through. You first need to know the issue (seems no one here has expirience with it). Two tools that are great are: 1) VOipMoitor and 2) sngrep |
| 16:08.54 | Dovid | sngrep is real easy to setup and you can see most issues with it right away |
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| 16:09.13 | Samot | Uhm. |
| 16:09.21 | Samot | Just looking at the call log is enough. |
| 16:09.26 | Samot | And making a live call. |
| 16:09.34 | Samot | That will tell you pretty much what you need to know. |
| 16:09.40 | eric_hill | Yep, that's the plan. Thanks all. |
| 16:09.43 | Dovid | tomato tomato |
| 16:10.00 | Samot | Crap like VoIPMonitor is just a resource hog that defeats its own purpose. |
| 16:10.04 | Dovid | i live on sngrep and voipmonitor |
| 16:10.31 | Dovid | i work for an ITSP so for us it's a must. lets us look back in the past rather then have to re-produce on demand |
| 16:10.41 | Samot | Yeah... |
| 16:10.44 | Samot | OK |
| 16:11.15 | Dovid | not to mention all the stats. When you have a local PBX with not a lot of calls its easy to trobleshoot from logs/cli. once you do thousands of current looking at sip codes makes things a lot easier |
| 16:11.17 | eric_hill | I've only used sngrep once, and never touched voipmonitor. But I've lived in Wireshark for days. |
| 16:12.01 | Dovid | eric_hill: sngrep is sort of like wireshark and you can run it via ssh vs wireshark u need to capture it and then download it locally or have it on the pbx's desktop |
| 16:12.14 | Samot | So does tcpdump |
| 16:12.19 | Samot | Or ngrep |
| 16:12.42 | Dovid | Samot: How is it a resource hog? I guess it depends on your call flow but I have it installed on every implementation |
| 16:12.56 | Samot | Well we had a guy in here the other night.. |
| 16:13.01 | Samot | Not even a week ago |
| 16:13.11 | Samot | 25% of his machine was taken up by VoIPMonitor. |
| 16:13.19 | Dovid | Samot: ngrep is in plain text, sngrep lets u see the call flow and spot issues a lot easier, with tcpdump again u still need to import to wireshark |
| 16:13.24 | Samot | The moment he turned it off, all his audio quality issues went away. |
| 16:13.32 | Samot | Dovid: I know. |
| 16:14.02 | Samot | I've done nothing but work at LECs or worked at/own/ran ITSPs. |
| 16:14.19 | Dovid | Samot: Depends on system specs. I have it running on all my systems including VPS's and I have no issues. the biggest hog is mysql which we put on another box. |
| 16:14.37 | Samot | 90% of my clients currently are VoIP/ITSP providers. |
| 16:15.05 | Dovid | Samot: as an ITSP it's critical to be able to look back instead of testing realtime when there is an issue. |
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| 16:15.14 | Samot | Yes and no. |
| 16:15.21 | Samot | Depends completely on the issue. |
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| 16:15.44 | Dovid | Samot: Mainly US Lecs or intl? |
| 16:15.50 | Samot | US |
| 16:16.18 | Dovid | ah ok. Intl is a whole other animal. |
| 16:16.29 | Samot | How so? |
| 16:16.42 | Samot | Intl doesn't mean anything |
| 16:16.52 | Samot | Since each country has their own rules. |
| 16:17.05 | Samot | Some may share those rules. |
| 16:17.27 | Dovid | My first talk ever was on troubleshooting: https://www.youtube.com/watch?v=XzSk1k5_Ma0&index=23&list=PLighc-2vlRgT3sWvW0RL7qyD2THenFIfZ |
| 16:17.41 | Samot | I've work with carriers in the UK and India. |
| 16:17.59 | Dovid | Samot: The subject of my talk this year: https://astricon2017.sched.com/event/BZWf/international-calling-cant-we-all-just-get-along |
| 16:18.14 | voipmonk | idiot |
| 16:19.08 | Dovid | When I say Intl I mean calling all over. We have clients that call every where you can think of. the crap the carriers pull off amazes me. For instance we had FAS with BICS. BICS is considerd a tier1. You would think inter-EU would have clean routes. |
| 16:19.29 | Samot | What do you mean "How can CallerID be unpredictable"? |
| 16:20.23 | Dovid | Samot: Like when we tried sending calls with L3 to France and they didnt pass it. We opened a ticket and L3 said "sorry we don't support caller ID on intl calls" - old school telco mentatlity |
| 16:20.36 | Dovid | or where people us GSM gateways to lower their cost. |
| 16:20.59 | Samot | When you say callerID are you talking name?! |
| 16:21.19 | Dovid | No Number. AFAIK only the US and canada have CNAM |
| 16:22.09 | Samot | What are they passing instead? |
| 16:22.16 | Samot | A source number has to be sent. |
| 16:22.35 | Dovid | Samot: We always send source. the issue is sometimes the CLI we send is not what the called party gets. |
| 16:22.53 | Samot | That's on *their* carrier. |
| 16:23.02 | Dovid | Samot: carriers like Verizon and BT wont even take a call if you dont send CLI |
| 16:23.23 | Dovid | Samot: Nope. If I use BICS and they use some crappy guy with a gateway in his baement then its on BICS |
| 16:23.28 | Samot | When a call comes into my network, I have a choice.. |
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| 16:23.52 | Samot | I can accept the CallerID being presented or I can change it. |
| 16:24.16 | Samot | Because I'm in the US that means I either keep the presented name or I dip for the name. |
| 16:24.16 | Dovid | Samot: I dont mean to block CLI. I mean the issues as to why even though I send CLI it wont make it to the other side |
| 16:24.40 | Samot | How I present that CallerID to my end users has no bearing on how you sent it to me. |
| 16:25.00 | Dovid | oops sorry. Your talking about incoming, I am talking about outgoing. I should make that more clear. |
| 16:25.07 | Samot | It's both |
| 16:25.26 | Samot | Your *outgoing* callerID format has no bearing on how I *present* it to the end user |
| 16:26.17 | Dovid | Not my format but I am going to dicuss the need for a good carrier and why caller ID is not always presented to the end user the way it should be |
| 16:26.26 | Samot | Again.. |
| 16:26.39 | Samot | The destination carrier has a say |
| 16:26.40 | Dovid | I feel like I am on a meryr go round |
| 16:26.47 | salviadud | Dovid, I got my callerid gripes from my local carrier, it's a worldwide problem IMO. |
| 16:26.57 | Samot | You can present your calledid however you want... |
| 16:27.01 | Samot | Once I have it.. |
| 16:27.12 | Samot | I can do what I want with it before sending it to the final destination |
| 16:27.18 | Dovid | Samot: correct. most of the time it isnt an issue. from my expirience 99% of CLI related issues are not the end carrier but the ones in middle creating issues |
| 16:27.40 | Samot | If I don't trust the name you present and I do a dip, the name I present could be different. |
| 16:28.00 | Dovid | Samot: I am not talkign about name, I am talking about number. |
| 16:28.16 | Samot | So they're sending calls with no number |
| 16:28.19 | Samot | Just "Private" |
| 16:28.30 | Dovid | either private or a local number shows up in place of what I send |
| 16:28.45 | Dovid | so its mostlikely a guy working out of his basement with a GSM gateway |
| 16:29.06 | Dovid | Samot: Do you still have contacts in India? |
| 16:30.18 | Samot | I could reach out if desired. |
| 16:30.28 | Dovid | Samot: I will PM |
| 16:30.32 | Samot | No. |
| 16:31.32 | Samot | I don't read unsolicited PMs |
| 16:32.16 | Dovid | Ok. We have requests every so often for Indian DID's. From what I understand the telco regulator is not allowing DID's to be sold. |
| 16:32.31 | Samot | Very possible. |
| 16:32.40 | Samot | I believe they still don't allow 5060 either. |
| 16:33.26 | Dovid | Ok. If they can get local numbers we have clients that want them |
| 16:33.39 | Samot | You have to have presence in India. |
| 16:33.42 | Dovid | currently the only option is to get a POTS line and then do a call forward |
| 16:34.04 | Samot | Unless you're going to open an office there and register with them.. |
| 16:34.12 | Samot | You're probably not going to get the DIDs. |
| 16:34.31 | Samot | Like I said, they block 5060. |
| 16:34.41 | Dovid | Ok. |
| 16:34.44 | salviadud | Samot, a bit off-topic, will you be attending AstriCon? |
| 16:34.53 | Samot | In order to provide US service to them I had to so a special setup for them. |
| 16:35.01 | Samot | Using 80 as their SIP Port |
| 16:35.13 | Samot | Because that was basically one of the only ports allowed for the outside |
| 16:35.28 | Samot | salviadud: No, I am not attending. |
| 16:35.38 | Dovid | Interesting. We have clients that connect to our servers in the US. All we did was set up an alt port (5744) and it works. using SIP over UDP, no encryption either. nothing that DPS wouldnt be able to see |
| 16:35.57 | Samot | The 5744 is an open port. |
| 16:36.13 | Samot | There is a reason they need non-standard or way outside ports for this. |
| 16:36.20 | Samot | Because the government blocks that crap |
| 16:36.59 | Dovid | yup. but from what I hear they are slowly deregulating it. lets see what happens. |
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| 17:47.05 | overyander | when I send a call to voicemail, it plays the users message (which i want) then plays the asterisk audio with instructions on how to leave a message (which I don't want). how can I set an individual mailbox to not play the instructions? |
| 17:49.47 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 17:50.37 | Samot | You set the s parameter. |
| 17:50.42 | [TK]D-Fender | <PROTECTED> |
| 17:50.50 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_VoiceMail |
| 17:51.44 | overyander | thanks! |
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| 18:28.45 | *** join/#asterisk andycol (~andycol@154.118.252.14) |
| 18:28.48 | andycol | Hi All |
| 18:28.58 | andycol | has anyone got the cisco 8845 ip phone working on asterisk here? |
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| 18:30.49 | Dovid | TK: Astricon this year? |
| 18:36.27 | [TK]D-Fender | Never been, not likely either |
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| 20:12.16 | salviadud | andycol, I've gotten similar phones to work |
| 20:12.53 | salviadud | just googled it. |
| 20:13.05 | salviadud | That thing looks like it actually works, nope sorry. |
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| 21:07.46 | cresl1n | sings the astricon song |
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| 21:09.14 | avb | hey guys. Since I have updated asterisk from 11 to 13, users are complaining about getting echo from client side on SIP. Looking familiar to someone? |
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| 22:01.21 | Samot | Can't say that it is. |
| 22:01.32 | Samot | I've been running 13 for a couple years now and I don't have those issues. |
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| 23:01.07 | avb | Samot: yeh, i have a feeling that its trunk related problem, not asterisk |
| 23:01.44 | avb | i cant imagine how SIP2SIP setup with directmedia enabled can cause echo problems |
| 23:02.08 | Samot | SIP2SIP where? |
| 23:02.22 | Samot | PBX <-> Endpoint? |
| 23:02.27 | Samot | PBX <-> PBX? |
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| 23:23.23 | tehgooch | Just wanted to share I got my dCAP certificate today :) |
| 23:23.41 | salviadud | Tell me before I google dCAP |
| 23:23.57 | Samot | Digium |
| 23:24.07 | tehgooch | Digium Certified Asterist Professional |
| 23:24.11 | tehgooch | Asterisk |
| 23:24.22 | tehgooch | Good thing spelling Asterisk wasn't on the test. |
| 23:24.58 | Samot | Meh. |
| 23:25.05 | Samot | No offense to Digium.. |
| 23:25.15 | Samot | I don't need a test to tell me things. |
| 23:25.19 | salviadud | Samot, do you consider it an easy cert? |
| 23:25.29 | Samot | I have no idea. |
| 23:25.33 | tehgooch | I feel the same way about certs TBH. Company paid for it all though. |
| 23:25.36 | Samot | I've never really looked into it. |
| 23:25.41 | Samot | Yes. |
| 23:25.44 | Samot | Companies do that |
| 23:25.48 | salviadud | I'm not certified either. |
| 23:25.54 | tehgooch | The hardest part about the test was the time limit. |
| 23:25.57 | Samot | Because certs add value. |
| 23:26.08 | salviadud | How much time did you have? |
| 23:26.25 | tehgooch | 90min for practical and 90min for written. |
| 23:26.42 | tehgooch | IIRC |
| 23:27.17 | tehgooch | Spinning up a fully functional phone system in 90 minutes was a little challenging. |
| 23:27.35 | salviadud | Fully functional with DAHDI? |
| 23:27.36 | Samot | I guess it depends on how basic or not it is. |
| 23:27.58 | salviadud | Or does that not matter? |
| 23:27.59 | tehgooch | DAHDI and PJSIP |
| 23:28.21 | Samot | Hrm. |
| 23:28.30 | Samot | What do they recommend for PJSIP prep? |
| 23:28.31 | salviadud | I would find that challenging since I haven't used pjsip yet. |
| 23:28.44 | Samot | Considering Asterisk 4th Ed. doesn't cover it |
| 23:29.04 | tehgooch | They had a 5 day course beforehand. |
| 23:29.18 | tehgooch | Which covered it all. I had never touched DAHDI before then. |
| 23:29.30 | Samot | And you probably never will. |
| 23:29.31 | tehgooch | Been using PJSIP on my personal PBX for months. |
| 23:29.39 | salviadud | Samot, I beg to differ |
| 23:29.41 | salviadud | You never know... |
| 23:29.48 | salviadud | I used DAHDI |
| 23:29.54 | Samot | Yeah. |
| 23:29.57 | tehgooch | Yeah my company doesn't touch analog or PRI. |
| 23:30.00 | Samot | Some people do. |
| 23:30.01 | Samot | ^^ |
| 23:30.09 | tehgooch | Except for when they do LOL |
| 23:30.24 | tehgooch | We usually use an appliance for that though. |
| 23:30.51 | tehgooch | Because nobody here really knows fuck all about it. |
| 23:31.01 | Samot | Not even Telco's are using TDM/copper for PRIs. |
| 23:31.25 | Samot | Well, depends on the service. Like in most cases it has to be asked for. |
| 23:31.36 | Samot | "No, a real PRI circuit." |
| 23:31.46 | tehgooch | One guy in the course uses exclusively PRIs. |
| 23:32.08 | tehgooch | He works in the radio industry though. |
| 23:32.16 | Samot | Right. |
| 23:32.21 | Samot | Certain places should have PRIs. |
| 23:32.26 | Samot | Either primary or backup. |
| 23:32.33 | Samot | Hospitals. |
| 23:32.45 | Samot | Emergency Services |
| 23:32.49 | tehgooch | We only use PRIs in medical applications and if the customer has a specific need. |
| 23:32.57 | Samot | Right |
| 23:32.59 | tehgooch | e.g. retirement homes |
| 23:33.06 | Samot | Because those circuits have SLAs. |
| 23:33.09 | Samot | MTTRs |
| 23:33.38 | tehgooch | Even then it's usually only as a backup. |
| 23:33.40 | Samot | Generally 4 hours or less. |
| 23:34.09 | Samot | Yes, because PRIs are still considered "mission-critical" |
| 23:34.32 | tehgooch | Yeah basically if VoIP fails it has to use *something* |
| 23:34.56 | Samot | PRIs are not going anywhere anytime soon overall. |
| 23:35.16 | Samot | Because certain industries have requirements that VoIP cannot meet. |
| 23:35.48 | Samot | Not having phones die when the Internet does. |
| 23:36.10 | tehgooch | In some cases we get around that by having fibre with a tight SLA. |
| 23:36.27 | Samot | DSL, Fiber those pass some level of mustard. |
| 23:36.35 | Samot | Because they are *dedicated* line. |
| 23:36.37 | Samot | Because they are *dedicated* lines. |
| 23:36.42 | Samot | Comcast is not. |
| 23:37.05 | tehgooch | eh, DSL can be hit or miss. Most places that can get DSL but not fibre are rural where the dsl is shite |
| 23:37.19 | tehgooch | at least in Canada |
| 23:37.24 | Samot | But it's still *dedicated* |
| 23:37.33 | tehgooch | dedicate shite is still shite |
| 23:37.34 | Samot | It's still a pair of wires on a post |
| 23:37.51 | Samot | It's why they can have tight SLAs. |
| 23:38.12 | Samot | Because for the most part it's going to be in *your* transport connection. |
| 23:38.18 | tehgooch | I mostly agree, dunny why I'm arguing. |
| 23:38.18 | Samot | Unlike Comcast.. |
| 23:38.22 | Samot | Or Cogeco. |
| 23:38.33 | Samot | Where everyone in an area shares a node. |
| 23:38.42 | Samot | So if that node goes down, everyone is down. |
| 23:39.07 | Samot | They just can't swap a fire stations pairs for a working pair |
| 23:39.47 | Samot | DSL, Fiber, PRI, T1..all that can be fixed per location. |
| 23:40.49 | Samot | But as far as certs go... |
| 23:41.12 | Samot | I've spent my entire career not having certs but cleaning up the messes of those with certs. |
| 23:43.14 | Samot | Things like the dCAP, to me, are even worse overall. Sure you've got your dCAP but it's only going to help you in places where Asterisk is used. |
| 23:44.19 | tehgooch | Yeah I don't really disagree. |
| 23:44.35 | Samot | The CLEC I was at had an entire cert/education path they laid out for me.. |
| 23:44.54 | Samot | Most of it was complete garbage and a waste of my time. |
| 23:45.14 | Samot | The whole reason they wanted it.. |
| 23:45.17 | Samot | Funding. |
| 23:45.24 | Samot | More Certs = More Value |
| 23:45.26 | tehgooch | I worked with multiple people with certs and they were all useless. |
| 23:45.53 | Samot | Certs are good for paper. |
| 23:45.57 | tehgooch | People who had certs and would tell you all about it that is. |
| 23:46.00 | Samot | It makes things look good on paper. |
| 23:46.23 | tehgooch | I've worked with people who had tons of certs and you'd never know. |
| 23:47.26 | Samot | I just blew away an entire QoS plan an MSP proposed.. |
| 23:47.36 | Samot | With their "certified network engineers" |
| 23:48.03 | Samot | In one paragraph because they didn't account for one minor issue. |
| 23:48.15 | tehgooch | My manager really just wants someone who isn't a complete buffoon so he can fire the guy with all the certs lol. |
| 23:48.26 | Samot | There wasn't a core router in place. Two networks were plugging straight into the Fiber ports. |
| 23:49.02 | Samot | Four lines made their 2 page proposal and them look like idiots. |
| 23:49.49 | tehgooch | sounds like fun |
| 23:50.35 | Samot | A cert means you had a good test day |
| 23:50.39 | Samot | That's it. |
| 23:50.48 | Samot | Unless you got the actual skills to back it up.. |
| 23:50.51 | Samot | It's paper. |
| 23:50.58 | tehgooch | good talk |
| 23:51.27 | Samot | My scale is simple.. |
| 23:51.42 | Samot | The more you brag about your certs, the less you actually know. |
| 23:52.04 | tehgooch | agreed. i was just happy i didn't choke and wanted to share :) |
| 23:52.15 | Samot | Cool that you passed... |