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| 01:24.06 | bsdice | hey my Asterisk does fax |
| 01:24.17 | bsdice | using iaxmodem and hylafax though |
| 01:26.17 | bsdice | because many VOIP providers do not support T.38 |
| 01:28.19 | sy2up | my asterisk does fax as well, over a SIP trunk that doesn't support T.38. Its awesome. Probably 3-5% of faxes fail to send on the first try though. Luckily the fax machine auto-retries 3 times. |
| 01:28.32 | sy2up | everyone told me it wouldn't work, but who's the one sending faxes now... |
| 01:31.15 | ChannelZ-Wk | old people |
| 01:31.42 | [TK]D-Fender | "the one" <- accurate count |
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| 01:48.18 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 01:48.48 | jeffspeff | setting up odbc for mysql. should i be using odbc.ini or odbcinst.ini? |
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| 01:53.14 | pabelanger | jeffspeff: both |
| 01:53.32 | jeffspeff | i see that now in the book. thanks pabelanger |
| 01:53.45 | jeffspeff | the book is awesome. :) |
| 01:53.56 | pabelanger | jeffspeff: send monies to leifmadsen |
| 01:54.05 | pabelanger | and russellb |
| 01:54.16 | jeffspeff | i should. is there a link for that? |
| 01:54.20 | pabelanger | ~book |
| 01:54.20 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 01:54.24 | pabelanger | ~buybook |
| 01:54.24 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
| 01:55.28 | jeffspeff | ah, i thought you were referring to donations. i bought the 3rd edition. using a .pdf for 4th edition for now. monies are a little high in demand. |
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| 01:58.14 | pabelanger | leifmadsen: setup bitcoin for donations |
| 01:59.13 | jeffspeff | i'm not sure using bitcoin in the US is a great idea. one day i read that it's ok, then the next people are arrested for laundering money with it or something. |
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| 02:02.03 | pabelanger | old news |
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| 02:43.27 | leifmadsen | pabelanger: jeffspeff: I have a public wallet already -- 12K3GmKK4gVvVxMzJGhGhz2gff7hVo6Usr |
| 02:44.14 | leifmadsen | pabelanger: btw you owe me mBTC for winning the bet about russellb's kid |
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| 03:13.03 | chare | so whats the latest best voip setup |
| 03:13.12 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 03:13.22 | [TK]D-Fender | Same as last week |
| 03:13.26 | [TK]D-Fender | Probably same as next week |
| 03:13.37 | [TK]D-Fender | And the next several dozen after that |
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| 03:19.54 | chare | so no news about google hangouts? |
| 03:20.54 | [TK]D-Fender | nope |
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| 04:18.42 | monsterco | Hi everyone - Is there anyway to monitor Aastra phones with something like Nagios? or it's not feasible and best to just monitor Asterisk? |
| 04:20.03 | WIMPy | Asterisk can monitor it's peers. ("qualify") |
| 04:23.56 | Penguin | Asterisk can monitor "it is" peers? |
| 04:24.33 | WIMPy | Yes, it can detect it is its peer as well. |
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| 07:18.32 | vedic | Is there any open source VoiceXML engine that can be integrated with Asterisk? I see there are many paid commercial solutions and they are very expensive |
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| 07:55.10 | liquidamber | anyone know if app_swift works with asterisk 1.8.20 |
| 08:17.38 | liquidamber | or if it works with anything, ever... i'm surprised cepstral recommends it |
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| 08:38.15 | skrusty | morning |
| 08:38.38 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 08:39.08 | snadge | any of you guys had any experience with A10 networks AX or SoftAX load balancers with asterisk? :p |
| 08:41.08 | snadge | such a strange thing.. if anybody does, they certainly don't talk about it.. load balancing sip.. its like a black art |
| 08:42.23 | snadge | i didn't realise when I signed up for this job, that it would be so complicated ;) |
| 08:43.27 | liquidamber | a lot of SIP / asterisk stuff is kindof new territory it seems like |
| 08:44.12 | snadge | well asterisk in itself is complicated enough |
| 08:44.21 | liquidamber | even the "big" providers are kindof crackerjacked together |
| 08:44.26 | snadge | and same with the SIP protocol with UDP |
| 08:44.50 | liquidamber | ringcentral uses kamailio |
| 08:45.31 | snadge | right.. i was about to mention that.. people do use freeswitch, and opensip / kamailio |
| 08:46.00 | snadge | but these arn't bolt in solutions.. you cant just take an existing asterisk setup, and magically expect to integrate any of the above with it |
| 08:46.07 | liquidamber | but really its like, pay me 200k a year to bandaid pretend this shit is OK until something catatrophic happens |
| 08:46.10 | Faustov | hmm, on the other hand, what can be so difficult to load-balance something that is session-based? |
| 08:46.12 | snadge | it is not a bolt in solution |
| 08:46.42 | liquidamber | what's the issue anyway, you have a call center on asterisk? |
| 08:46.43 | snadge | if you design your network with that in mind from the start.. then yes.. you can expect to have some degree of success with it |
| 08:46.50 | snadge | absolutely |
| 08:47.00 | snadge | and it is load balanced using round robin dns |
| 08:47.12 | liquidamber | inbound calls only / mostly? |
| 08:47.19 | liquidamber | (i dont know shit, just curious)( |
| 08:47.32 | snadge | well the bread and butter is hosted pbx.. and outbound |
| 08:47.39 | Faustov | well, RR dns seems like the most robust approach |
| 08:47.44 | snadge | but yes, theres plenty of inbound as well.. there is no money in that though |
| 08:48.39 | snadge | i cant remember the stats of the top of my head.. but its something like a few million minutes a month |
| 08:48.43 | liquidamber | im really amazed that there is money for small providers with that. just seems like endless support calls, unhappy customers, unmet expectations, etc. but, i digress |
| 08:48.57 | snadge | our biggest competitor.. is something like 10 times that |
| 08:50.14 | liquidamber | i guess the minutes and bandwidth cost almost nothing and everything else is profit |
| 08:50.29 | snadge | that.. and we're a relatively small company |
| 08:50.53 | snadge | dont get me wrong.. there are voip providers that are smaller.. we just bought one.. eg.. 3 people |
| 08:51.17 | liquidamber | what happens when verizon and AT&T collude to shit on voip is my question |
| 08:51.19 | snadge | a sales guy, an investor, and a tech guy |
| 08:51.41 | snadge | but we actually have a sales guy.. a support team.. and now two techs.. wow ;) |
| 08:52.05 | snadge | and have been around for about 6 years or something.. im the new tech |
| 08:52.28 | liquidamber | whats your edge hardware? |
| 08:52.39 | liquidamber | ive heard a lot of bigger providers like juniper |
| 08:53.05 | snadge | right.. and my boss, he likes juniper.. he wants to go down that path.. but the other tech.. he is a full on linux nazi ;) |
| 08:53.15 | liquidamber | oh he wants to firewall everything with iptables, great |
| 08:53.38 | liquidamber | so, no SBC? |
| 08:53.47 | snadge | so without wanting to go into too much detail about our setup.. lets just say our primary router, runs linux.. its actually an impressive build, I will give him that.. and i personally find it difficult to fault it |
| 08:53.53 | snadge | most routers are woefully underpowered |
| 08:54.08 | liquidamber | hmm wow |
| 08:54.29 | liquidamber | their CPU may be but once the calculation is complete the routing is done in silicon |
| 08:54.46 | liquidamber | but, it's usually not an issue for that kind of setup |
| 08:54.55 | snadge | right.. but out routing table isn't exactly small |
| 08:55.00 | liquidamber | ah |
| 08:55.17 | snadge | i think a lot of cpu is taken up by checking against quite significant block lists etc.. and its all dynamic |
| 08:55.20 | liquidamber | well, thats terrifying because i wouldnt know where to even start looking at it |
| 08:55.27 | liquidamber | but, it sounds interesting |
| 08:55.47 | snadge | yeah.. the routers are aging by todays standards.. but we're consistently pushing above 50mbit |
| 08:56.03 | liquidamber | you have already identified a problem though, you have 1 guy who understands some thing he built that could or could not be a problem |
| 08:56.23 | snadge | well to be honest.. given the load that they're under etc.. they're not actually the weak point |
| 08:56.51 | snadge | we can just keep adding asterisk servers.. to handle more calls |
| 08:57.06 | liquidamber | theyre all physical asterisk servers |
| 08:57.07 | liquidamber | ? |
| 08:57.11 | snadge | but using round robin dns.. probably not the best approach |
| 08:58.00 | snadge | we're not having issues with system load, or anything like that currently.. the calls are quite evenly spread.. and theres a significant amount of head room |
| 08:58.47 | snadge | one issue is people who wish to run diallers.. will place one of the servers under considerable load, and theres other issues to content with like the database.. which has to bill each call etc |
| 08:59.33 | snadge | they may as well be physical, but we do use virtualization yes |
| 08:59.59 | snadge | to the best of my knowledge, that isnt a problem though.. we dont have issues with call quality, drop outs etc.. or overutilisation issues |
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| 09:02.52 | liquidamber | i thought asterisk had scalability issues |
| 09:02.58 | snadge | oh it definitely does |
| 09:03.01 | liquidamber | which is why freeswitch was better for multi tenancy |
| 09:03.33 | liquidamber | again im a novice thoug just curious |
| 09:03.53 | snadge | freeswitch is somewhere in between kamailio/opensip and asterisk |
| 09:04.02 | snadge | its not quite a router, and its not quite a full blown pbx |
| 09:04.16 | snadge | asterisk is definitely a more mature product |
| 09:04.23 | liquidamber | it's exploitable as hell though especially with add ons |
| 09:04.58 | snadge | i also think asterisk 12 is addressing some of the scalability issues.. but really.. as long as you know what your ceiling is, and you keep your load below that |
| 09:05.00 | snadge | and spread it out |
| 09:05.03 | snadge | its not really an issue |
| 09:05.08 | liquidamber | what are like, 2600 / kazoo / twilio doing |
| 09:06.05 | liquidamber | or even voip.ms |
| 09:06.10 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 09:06.29 | snadge | i dont know.. im fairly new to this.. and my boss, hes actually quite familiar with the australian voip market.. which is obviously a drop in the ocean with regards to voip worldwide |
| 09:06.46 | liquidamber | i just ask because they have to have run into the same problems |
| 09:07.04 | snadge | i think the simplest answer is.. the really big guys, don't use asterisk |
| 09:07.15 | liquidamber | theyre in the middle though or grew really quickly and they do use asterisk |
| 09:07.34 | snadge | so i guess the question becomes.. who are the largest users of asterisk |
| 09:07.38 | liquidamber | or at least, they did and migrated, which is still relevant |
| 09:07.47 | snadge | exactly |
| 09:09.10 | snadge | one of the issues is .. when you get to that middle size, where you're starting to outgrow asterisk.. you dont really have the luxury to just throw it away, and use something else |
| 09:09.19 | snadge | because its going to cause a significant impact to your customers |
| 09:09.23 | snadge | no matter how you try to engineer it |
| 09:09.34 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.204) |
| 09:09.52 | snadge | the boss tried kamailio.. and im sure its great if you deploy new customers on that, and test it all etc |
| 09:10.24 | snadge | but you cant just plug it in, and hope for the best.. without expecting all hell to break loose |
| 09:10.39 | liquidamber | well i know voip.ms just makes their customers pick a geographical location and puts their POP on only one server |
| 09:11.07 | liquidamber | and i think ringcentral does something similar, but via phone provisioning |
| 09:12.40 | snadge | right.. the issue becomes.. what happens when you exceed any particular servers maximum call throughput |
| 09:13.03 | snadge | obviously you can try to identify the larger volume customers and manually move them to another server |
| 09:13.28 | snadge | but this isnt really a sustainable option, growing into much higher volumes |
| 09:13.31 | liquidamber | so you're seeing high CPU utilization |
| 09:13.33 | liquidamber | essentially |
| 09:13.42 | liquidamber | because of transcoding or what |
| 09:14.11 | snadge | i would actually need to talk to my boss and the other engineer about that specifically.. but my understanding is its not necessarily a cpu limitation |
| 09:14.31 | snadge | you can run out of source ports or something like that.. its the nature of how UDP works |
| 09:14.40 | snadge | and the fact that theres only 64000 available ports |
| 09:14.47 | liquidamber | right |
| 09:14.57 | liquidamber | you mentioned autodialers... just throttle them :P |
| 09:14.57 | snadge | and virtualisation is actually a bandaid for that |
| 09:15.14 | snadge | since you can just create more vms |
| 09:15.24 | snadge | and each vm gives you another 64000 ports ;) |
| 09:15.43 | snadge | but round robin dns has obvious limitations for load balancing |
| 09:16.29 | snadge | as just one example.. a dialer might only resolve the hostname once.. and then say.. here you go, have 20,000 calls.. 1 every millisecond |
| 09:16.57 | liquidamber | why not throttle the peer |
| 09:17.06 | snadge | or.. customers might hard code the ip address of one server into their pbx.. or devices.. whatever.. and not the hostname |
| 09:17.22 | liquidamber | maybe something where you can decline registrations |
| 09:17.26 | snadge | so what you'll see then.. is one server loading up more than the others |
| 09:17.31 | liquidamber | can you redirect or something |
| 09:17.55 | snadge | right.. thats what we were looking into.. and citrix has a product called netscaler, which can actually do something along those lines |
| 09:18.13 | snadge | and after asterisk 1.6, you can have a shared registration database |
| 09:18.15 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:18.19 | liquidamber | yeah a hardware load balancer might be able to do something like that, but i thikn the SIP aware ones are all called SBCs |
| 09:18.23 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:18.46 | snadge | so what that means is.. a client might only register to 1 server in a cluster.. but because that cluster uses a shared database |
| 09:19.01 | snadge | when the load balancer redirects a call to one of the other "nodes" .. the server wont just say.. bugger off, you haven't registered to me.. go away |
| 09:19.09 | liquidamber | very cool |
| 09:19.25 | liquidamber | if digium were smart they'd make an SBC :P |
| 09:19.37 | snadge | apparently AX is more sophisticated than that |
| 09:19.44 | snadge | which is from A10 networks |
| 09:19.55 | snadge | but its so complicated, I haven't been able to figure out how to get it to work ;) |
| 09:20.29 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 09:20.49 | liquidamber | so really you need something to drop inbetween your wan and your asterisk clusters with minimal impact |
| 09:21.05 | snadge | exactly.. its unfortunately non trivial |
| 09:21.39 | snadge | ax is interesting though.. you have 3 interfaces.. a management interface for configuring the load balancer itself, an input, and an output interface |
| 09:21.53 | snadge | in routed mode.. the idea is, you plug each interface into a switch, and use vlans for each one |
| 09:23.03 | snadge | apparently in this configuration.. it means our asterisk servers can still have public ips, and be accessible directly.. or they can be accessed via the load balancer |
| 09:23.18 | liquidamber | oh, that is neat |
| 09:23.22 | snadge | how this works, is beyond both my understanding.. and the understanding of the vastly more experienced engineer that I work with ;) |
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| 09:23.27 | liquidamber | if it works like you say |
| 09:23.50 | liquidamber | sounds like the appliance would juggle the translations |
| 09:24.23 | snadge | right.. i've sent the a10 guy an embarassingly simple diagram of our network setup.. because I said i wanted to see how the AX works itself |
| 09:24.31 | snadge | and didnt want to complicate it with irrelevant details |
| 09:25.22 | liquidamber | well whatever is going on there is a much worse setup somewhere else |
| 09:25.28 | snadge | load balancing UDP is complex.. adding sip to that, takes it to a level that is beyond most of your http/ssl load balancer guys understanding |
| 09:26.18 | snadge | most people just want to scale web applications.. or things which work very similar to that.. at least with tcp, you have a concept of a session |
| 09:26.22 | liquidamber | yeah you kinda have to have someone who knows SIP, RTP, UDP, asterisk, and whatever else you have in the mix |
| 09:26.32 | liquidamber | otherwise they focus on one part |
| 09:27.09 | snadge | right.. and many of these setups have the luxury of being able to say.. screw the existing customers... lets start from scratch |
| 09:27.12 | liquidamber | i was at a placea year ago that was looking for someone to build their infrastructure, they never found one |
| 09:27.19 | liquidamber | they ended up partnering and reselling some other shit |
| 09:27.52 | snadge | when you can control exactly what connects to the input.. you have a much easier time of it.. things like skype, and google voice etc.. impressive, but not really |
| 09:27.56 | snadge | they are very tightly controlled |
| 09:28.11 | liquidamber | ive heard facetime is kindof interesting |
| 09:28.19 | liquidamber | uses STUN, SIP and RTP |
| 09:28.44 | snadge | interesting.. but apple only |
| 09:30.29 | snadge | there are so many sip devices out there.. being able to let any of them connect and do their thing, is difficult.. asterisk is actually incredibly good at this |
| 09:30.48 | snadge | other things are vastly more scalable.. but less flexible |
| 09:31.56 | snadge | the fact that freeswitch can do 200,000 calls on a dual core processor.. is largely irrelevant, if it wont talk to an outdated Yealink, or someones obsolete FreePBX system running on a pc under someones desk.. for the last 5 years |
| 09:32.26 | snadge | its horrible to think about it.. but thats the sort of stuff that we deal with, as a reality, on a daily basis |
| 09:33.29 | liquidamber | nah thats awesome :) |
| 09:33.51 | liquidamber | amazing that they're not compromised and toll frauded, but i guess that means they were configured competently |
| 09:34.27 | liquidamber | and last time i checked asterisk is the only one with chan_unistim |
| 09:35.29 | snadge | well.. we have systems in place to detect fraud for that reason.. some of these people just have no idea |
| 09:36.01 | snadge | others have set up a pretty tight firewall.. and dont allow remote extension logins.. or have very complex passwords etc |
| 09:36.21 | liquidamber | company i worked for had freepbx open to the internet, a skid found a vulnerability and then closed it :P |
| 09:36.49 | liquidamber | i found it by trying to exploit the same bug |
| 09:36.54 | liquidamber | but it had some sassy error message |
| 09:37.07 | liquidamber | point being, theres tons of freepbx out there that are compromised like you say... |
| 09:37.44 | snadge | freepbx just got hit by another XSS vulnerability.. but the only part of that, which should be open to the internet.. is port 5060 udp |
| 09:38.04 | snadge | and even then, by default now, it has fail2ban |
| 09:38.20 | snadge | so repeated failed auths will be rewarded with DROP |
| 09:38.28 | liquidamber | i just hate running iptables on my edge |
| 09:39.26 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 09:42.22 | snadge | its friday evening here.. and im talking about work related stuff.. granted im drinking beer.. at least thats something ;) |
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| 09:44.24 | liquidamber | 2am friday here, i'm drinking a beer, have work at 2pm tomorrow |
| 09:50.22 | Zogot | its very interesting though. im new to the whole VOIP stuff at the moment. starting a new job with it next month |
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| 10:05.56 | snadge | i wish you all the best Zogot ;) |
| 10:06.58 | snadge | ive had my job for a few months now.. and only today, I looked at the SIP rfc |
| 10:08.06 | snadge | my first thought was.. i wonder how easy it is to crack sip passwords, which lead me to an article about it :P |
| 10:08.37 | snadge | best practise.. at least 8 chars, 10 preferably.. alphanum, mixed case, ascii |
| 10:08.56 | snadge | otherwise your password may as well be plaintext |
| 10:08.56 | liquidamber | autoprovisioning is probably the worst with that |
| 10:09.41 | TSM2 | snadge: passwords are not sent as plaintext with SIP |
| 10:10.10 | snadge | i know this. but my point is.. if your password is less than 6 chars, and alpha only.. it may as well be sent in plaintext |
| 10:10.39 | snadge | they are simply md5 hashed with a token |
| 10:11.02 | TSM2 | true being short will make it easier but there is still the brute force required, you can offset the brute force speed by limiting the SIP packet rate |
| 10:11.16 | TSM2 | but a good password will always help |
| 10:12.37 | TSM2 | time to crack increases non-linerly due to the latancy |
| 10:14.35 | liquidamber | what if your business depends on phones that provision themselves |
| 10:15.00 | TSM2 | liquidamber: ?? |
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| 10:15.40 | liquidamber | voip provider says, get a phone from us, plug it in, and it will work |
| 10:16.13 | liquidamber | you set the phone up online and it is pointed to their provisioning server, where it downloads configuration files, usually in plain text, with credentials |
| 10:16.25 | liquidamber | this is pretty common |
| 10:16.57 | TSM2 | liquidamber: depends if they are using FTPS/HTTPS, also some phones support encrypted config files like the polys, ok granted not all will use it |
| 10:17.07 | liquidamber | many dont use it :P |
| 10:17.35 | liquidamber | just saying.. it's a shit show out there |
| 10:17.58 | TSM2 | liquidamber: you are assuming that there is a MITM attach, but also even if there was not, they can hide the files from directory listings, they could also put fail2ban checking for all attempts to download configfiles that do not exist |
| 10:18.45 | liquidamber | im not assuming mitm. 0004f2******-phone.cfg |
| 10:19.48 | TSM2 | well thats if they only supply polys |
| 10:20.02 | snadge | TSM2, its worse than that.. brute forcing by attempting authentication is one thing.. and fail2ban significantly addresses that |
| 10:20.21 | snadge | what im talking about is a mitm.. where the attacker literally logs both the challenge and response packets from the legitimate client and server |
| 10:20.31 | snadge | and then reverse engineers a non complex password from that |
| 10:20.37 | liquidamber | you see a lot of SIP brute forcing also... |
| 10:20.46 | liquidamber | like, extensions |
| 10:21.03 | TSM2 | this is always a problem, comapneis have to start using the encrypted config files but they are a pain to manage |
| 10:21.06 | snadge | there is an excellent writeup about it online.. where a guy uses an amazon ec2 cloud as an example |
| 10:21.38 | snadge | well the passwords must be stored in plain text, or obfuscated.. and we're assuming for the moment, that its not possible for the attacker to get access to that |
| 10:22.09 | snadge | if they've got root inside your network.. you are screwed.. any company or application is completely vulnerable in that scenario |
| 10:22.25 | TSM2 | on polys the whole config gets encrypted if you go down that route, it requires the phone to be autoprovisioned in a set order to get the certificate onto the phone then download the config, tricky |
| 10:23.23 | TSM2 | if someone has root then you are screwed anyway so just protect against that as a first |
| 10:23.40 | snadge | https://www.sipsorcery.com/mainsite/Help/SIPPasswordSecurity |
| 10:24.33 | TSM2 | ive read that |
| 10:25.27 | snadge | im not sure how accurate those figures are.. but it does make a reasonably solid case |
| 10:27.37 | liquidamber | night |
| 10:27.56 | TSM2 | does sha support exist in sip auth yet? |
| 10:35.46 | snadge | no, but you can use tls with tcp |
| 10:35.55 | TSM2 | i know |
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| 10:36.15 | snadge | it would appear the solution is to make the passwords annoyingly complicated ;) |
| 10:36.30 | TSM2 | easiest way currently |
| 10:36.31 | snadge | when you consider that most people cut and paste them anyway.. it's not really that big of a deal |
| 10:37.51 | snadge | as a classic example.. i shouldn't say that this actually happened to protect those concerned ;) |
| 10:38.09 | snadge | someone has a pbx.. that gets pwned.. lets say.. two or three times in a row |
| 10:38.17 | snadge | complete reinstalls .. firewalls etc |
| 10:38.44 | snadge | the email account that the password details were being sent to.. was hacked |
| 10:40.02 | TSM2 | div |
| 10:41.18 | snadge | thats an obvious fail.. you can communicate passwords over the phone, or in an encrypted manner.. in practise, not everyone thinks to do something like that |
| 10:44.14 | snadge | theres are numerous social engineering techniques that will undoubtedly succeed in gaining access to.. plus just glaringly bad oversights and lapses in judgement, with regards to protecting critical infrastructure.. like power and water companies.. etc |
| 10:44.35 | snadge | best just not to worry :p |
| 10:50.00 | TSM2 | act in bliss without a care in the world |
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| 11:31.08 | enzo | Hello |
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| 11:35.24 | enzo | I'm upgrading an AGI perl to asterisk 1.8, and $AGI->exec('SetMusicOnHold', 'random'); is indicated as deprecated, I should use Set(CHANNEL(musicclass)=random), but I don't find how to put that in my AGI perl script, any idea? |
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| 11:45.25 | enzo | well it seems to be $AGI->exec('Set', 'CHANNEL(musicclass)=random'); |
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| 13:13.38 | As001 | Hell can someone point me to documentation where i can read what is fromnumber line in sip.conf and will it overwrite callerid that I set when I call via that trunk with fromnumber= |
| 13:14.03 | As001 | *Hello sorry |
| 13:15.00 | [TK]D-Fender | that field does not exist |
| 13:18.23 | As001 | strange... That was I thought. My voip provider said i need to put it in peer configuration but I can't find any info about what does this do. lol. |
| 13:18.23 | wdoekes | As001: you're looking for fromuser, and it will be used instead of the CALLERID(num) in the From: |
| 13:18.45 | wdoekes | you can use sendrpid to send out CLI info as well through a separate header |
| 13:18.48 | [TK]D-Fender | And that is ONE form for the other side to use as "callerid |
| 13:19.17 | [TK]D-Fender | The other is RPID |
| 13:19.57 | [TK]D-Fender | So set "sendrpid=yes" and "trustrpid=yes" and that will send it in an alternate set of headers in the INVITE |
| 13:20.06 | WIMPy | Or PAI |
| 13:20.23 | As001 | what is PAI ? |
| 13:20.52 | WIMPy | P-Asserted-Identity |
| 13:21.24 | As001 | ok my version is Asterisk 1.6.24 and I use CALLERID(num) and CALLERID(name) to set callerid. |
| 13:21.39 | [TK]D-Fender | that is not a version.... |
| 13:22.12 | [TK]D-Fender | And you should get off of whatever it is ASAP |
| 13:22.29 | As001 | 1.6.2.24 sorry |
| 13:22.48 | Chainsaw | As001: Where are you calling from? 2009? |
| 13:23.40 | Chainsaw | As001: I hope you're not being held hostage by some distribution package, at any rate. You need 1.8, and the latest 1.8 at that. |
| 13:24.54 | As001 | Well I had strange Retransmission timeout in 1.8.23 which I could not solve and ugly hangups after 6-7 seconds for incoming calls.I read somewhere on internet that someone had same trobule and revert back to 1.6 and it was ok. So I am trying that. |
| 13:25.39 | As001 | other side did not sent my ACK on 200 OK after 10 retransmissions and asterisk hangsup call. |
| 13:25.42 | Chainsaw | As001: Okay, good luck. It is no longer maintained, no longer secure and stopped being a good idea several years ago. |
| 13:26.43 | [TK]D-Fender | As001: we |
| 13:27.15 | [TK]D-Fender | As001: we're at 1.8.25 and that's in that branch.. you are 2 entire branches behind and using one that is no longer supported whatsoever |
| 13:28.16 | Chainsaw | And if you're going to have the disruption of moving away from 1.6, you might as well try 11 now. |
| 13:28.17 | As001 | Yes I am aware of that just want to see if those retransmission will stop. |
| 13:33.20 | mjordan | As001: Without knowing what is actually occurring, no one can accurately answer that question. |
| 13:34.52 | As001 | well call comes to me I send 200 OK and voip provider does not send ACK, I retransmit 1..10 times, no answer and 1.8.23. hangup call. I use nat=no everywhere, asterisk box is not behind nat, outgoing calls go fine. |
| 13:35.47 | [TK]D-Fender | As001: None of that proves things are set up right. You talk about settings but we don't see what's actually going on. |
| 13:36.04 | [TK]D-Fender | As001: Perhaps you've made any number of mistakes. |
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| 13:36.43 | [TK]D-Fender | As001: I highly recommend showing these comm issues in full when you encounter them so that others' eyes can verify if you've missed something |
| 13:37.29 | As001 | ok I can go back to 1.8 and pastebin sip debug when it occures. |
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| 13:38.10 | As001 | ok thanks for help |
| 13:38.33 | As001 | when I recompile I will come back here to show problem. |
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| 13:42.41 | vassilux | Hi alls , What is T2 hangup cause 59 ? |
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| 14:06.31 | WIMPy | Undefined. |
| 14:06.48 | WIMPy | Category service or option not available |
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| 14:29.05 | jeffter | hellp |
| 14:29.09 | jeffter | hello* |
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| 14:30.20 | jeffter | need help with a litle project please |
| 14:32.11 | pabelanger | ~ask |
| 14:32.11 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:33.02 | jeffter | okay, i am working in a projet with asterisk, kinda offtopic of the normal |
| 14:33.28 | jeffter | i tried calendaring, with google but just doesn't get it, i need something else |
| 14:34.17 | jeffter | so i've been told to work with an external calendar, meaning not integrated, the calendar exports an *.csv file that asterisk must read and then proceed to call |
| 14:34.17 | TSM2 | and |
| 14:34.54 | TSM2 | ical format, you can poll google calendar that way to get the latest things |
| 14:35.15 | jeffter | example: i need something like this, jon needs to walk the dog at 1pm, the csv file must have hist number and info about the event, so then i could make an automatic call with his number |
| 14:35.17 | pabelanger | ~book |
| 14:35.17 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:35.23 | pabelanger | jeffspeff, chapter 18 ^ |
| 14:35.49 | jeffter | the question is can asterisk integrate the file in dialplan' |
| 14:36.03 | TSM2 | not directly no |
| 14:36.25 | jeffter | yes i've tried google, i can't use google calendar because they already have a calendar that exports the csv files |
| 14:36.26 | TSM2 | you will need to hand it off to an external script ie PHP and then use AGI to handle call setup etc |
| 14:36.43 | [TK]D-Fender | yup |
| 14:36.47 | jeffter | yes, but i just can't get there |
| 14:37.24 | TSM2 | read the book, look at how dialplan.php works as its a good example on what can be done |
| 14:37.32 | TSM2 | or dialplan.agi i cant remember |
| 14:38.13 | jeffter | hum... |
| 14:38.27 | jeffter | sounds complicated, i only got the basis of asterisk |
| 14:38.35 | jeffter | but thanks for your help |
| 14:39.33 | pabelanger | jeffspeff, the link above gives you a working example using ical |
| 14:39.35 | [TK]D-Fender | You don't do scheduling in the dialplan.. you do that outside before you trigger the system to call out. |
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| 14:43.55 | jeffter | ya, but i cant change the format of the event file |
| 14:44.06 | jeffter | they got the software, i just need to integrate it with asterisk |
| 14:44.33 | jeffter | its csv exported so i can't go ical, or calldav, etcv |
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| 14:44.45 | jeffter | yes im reading the book |
| 14:44.52 | jeffter | for the best solution |
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| 14:46.29 | [TK]D-Fender | If you have CSV... do it YOURSELF. |
| 14:46.43 | pabelanger | Then you need to build some middleware, load the CSV into it, then notify asterisk |
| 14:46.44 | [TK]D-Fender | make your own script to read it and launch the calls |
| 14:46.56 | pabelanger | or setup ical server, local CSV into it, then connect asterisk to ical |
| 14:47.13 | pabelanger | Or, just find an existing dialer project for asterisk that supports CSV |
| 14:47.17 | pabelanger | or pay somebody to do it |
| 14:47.31 | pabelanger | lots of options |
| 14:48.18 | jeffter | pay is out of option xD |
| 14:48.26 | jeffter | the only option is to learn and do it myself |
| 14:48.37 | jeffter | which will take time but if its possible |
| 14:48.41 | jeffter | its okay |
| 14:49.02 | [TK]D-Fender | A lot of things are possible. Especially with more time |
| 14:49.46 | jeffter | my skills aren't in this area |
| 14:49.55 | jeffter | im learning asterisk for 2 months now |
| 14:50.08 | jeffter | thats why its going to take time |
| 14:51.21 | jeffter | actually |
| 14:51.29 | jeffter | ii can't find anything in the book |
| 14:51.34 | jeffter | about dialplan.php |
| 14:52.05 | jeffter | time for google |
| 14:52.28 | [TK]D-Fender | ho such thing |
| 14:52.30 | [TK]D-Fender | no* |
| 14:52.59 | [TK]D-Fender | You do your own script to load the csv and trigger the call-outs. |
| 14:53.17 | [TK]D-Fender | that is "CALL FILES", or "AMI ORIGINATE" |
| 14:53.26 | jeffter | yap |
| 14:53.38 | jeffter | i'm thinking of something like |
| 14:53.52 | jeffter | identifying the number, (its all i need from csv) |
| 14:53.55 | [TK]D-Fender | Those are Googleable. And that' jsut to initate the call. You still have to understand the dialplan in order to actual process those calls one initiated |
| 14:54.44 | jeffter | yes, il try my best |
| 14:55.10 | jeffter | hope it works, with the script to identy and generate temp call files |
| 14:55.14 | jeffter | so asterisk can execute |
| 14:55.54 | jeffter | the calls |
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| 15:04.51 | TSM2 | jeffter: /var/lib/asterisk/agi-bin/dialparties.agi |
| 15:06.16 | jeffspeff | pabelanger, i think you were calling out the wrong jeff. |
| 15:06.59 | jeffter | he is, noticed that other jeff |
| 15:08.12 | pabelanger | jeffspeff, doh |
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| 15:10.27 | jeffspeff | well, now that i'm here. i'm toying with multiple parkinglots. i set the [parkinglot_a], [parkinglot_b] etc. in features.conf. the value set within each of those for context= is what i need to include in the dialplan right? or do i also need to make a matching context within the dailplan and use Park()? |
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| 15:30.40 | Katty | file: what if i don't want a cookie |
| 15:30.47 | Katty | file: what if i want moar schnitzel |
| 15:31.53 | carrar | You need some sauerkraut |
| 15:32.17 | carrar | hahah google |
| 15:32.20 | carrar | so funny |
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| 16:29.01 | jeffter | do i require any php dependencies to run php-agi scripts? |
| 16:41.28 | [TK]D-Fender | Depends on your use of the term "php-agi" |
| 16:42.01 | [TK]D-Fender | So far you havent mentioned what you are even going to do in these calls, so we don't know if AGI is even a need. |
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| 16:51.49 | jeffspeff | well, now that i'm here. i'm toying with multiple parkinglots. i set the [parkinglot_a], [parkinglot_b] etc. in features.conf. the value set within each of those for context= is what i need to include in the dialplan right? or do i also need to make a matching context within the dailplan and use Park()? |
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| 17:33.37 | newtonr | jeffspeff, yeah you need to include the "context=" contexts in your dialplan so that the extensions within can be dialed |
| 17:36.35 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 17:38.41 | cvance | I am having trouble with some calls coming in. Some calls work fine and I have guest calls disabled. I have some calls coming in with a cid of edicentral.net. It hits my call queue, but when the pick up the phone, it does not connect. The same calls then goes on to ring a second line and can still not be answered. |
| 17:40.02 | cvance | When I look through the log, the caller id is edicentral and a huge amount of whitespace: edicentral.net <multiple lines of whitespcae> Clic") |
| 17:40.42 | cvance | The phones are spa504g phones, has anyone had problems with faulty caller id passing in too many characters and preventing a phone from answering the call? |
| 17:42.20 | newtonr | cvance, I found a bug in the older SPA942 series where sending some goofy callerid to the phone would crash it. I reported it to Cisco/LinkSys but they never responded. I wonder if there is still something similar. |
| 17:43.06 | newtonr | cvance, what version of Asterisk, and have you looked at a packet capture to see what the queue member phone sends to Asterisk when it tries to answer? |
| 17:43.53 | cvance | newtonr, yeah it doesn't crash the phone for me, but kinda acts as a denial of service because as soon as someone tries to answer the call, the queue spawns another ring and takes up another line etc, until the 4 lines are maxed out. |
| 17:45.09 | newtonr | cvance, strange. Step 1 is to look at an Asterisk log and see what verbose and debug say when the phone tries to answer, and to look at a packet capture along with that to see what SIP messages the phone is sending to Asterisk. |
| 17:45.27 | newtonr | cvance, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 17:46.06 | cvance | Yes, I will do that, this is above my expertise, my asterisk version is old 1.6.2.24 running asterisknow |
| 17:46.33 | newtonr | oh, yuck yeah that version could have quite a few bugs in it as it has been end of life for a while |
| 17:47.17 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 17:47.20 | cvance | Do you have a asterisk appliance that you like, I have been meaning to upgrade |
| 17:47.39 | cvance | I first used asterisknow, but tried pbxiaf and elastix |
| 17:48.03 | cvance | I guess they are distributions but in your opinion do you have one that you like? |
| 17:50.10 | newtonr | The latest AsteriskNOW is good http://www.asterisk.org/downloads/asterisknow, or else many use the FreePBX distro http://www.freepbx.org/freepbx-distro |
| 17:50.26 | newtonr | Are you using the FreePBX GUI to manage your system right now? |
| 17:53.50 | newtonr | The others are fine too. Everyone likes them for different reasons. I'm a Digium guy and I don't administrate an Asterisk system so I'm not the best to ask for a personal opinion on which distro is best for you. :) |
| 17:58.35 | cvance | thanks! I'll probably implement the freepbx distro as I've heard good things |
| 17:58.41 | cvance | :D |
| 18:00.09 | *** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
| 18:02.13 | Penguin | barks |
| 18:07.07 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 18:08.18 | newtonr | cvance, welcome. also note that the #freepbx IRC channel is the best place to Asterisk questions if you happen to be administrating the system with FreePBX, as the answers often depend on the context of how you are managing your config files (writing by hand, or using a GUI) |
| 18:11.25 | cvance | yes, I've got a bit of a frankenstein, partial gui management with additions in the _additional.conf files :P, I'll reroll our phone system and try to follow best practices more closely next time. |
| 18:12.45 | newtonr | well, making custom additions is fine, as long as you know what you are doing. That is up to you to decide :D |
| 18:15.06 | Penguin | digiv: You really don't need to change your nick each time you leave the keyboard. IRC has a nice /away feature which will show anyone interested whether or not you are here. The rest of us don't care, don't need to see that you are away, and now your actions have taken up even more bandwidth making me explain this to you. |
| 18:15.55 | Penguin | newtonr: The problem with that is most people can't accurately assess if they know what they are doing. They think they do and end up ruining something. |
| 18:16.06 | *** part/#asterisk digiv (~digiv@as1.si.umich.edu) |
| 18:18.23 | newtonr | Penguin, It is even harder to accurately assess others, as you don't know what they do or don't know without a lot of investigation. |
| 18:18.24 | cvance | okay, newtonr thanks for the help and advice, take care I'm off |
| 18:18.35 | *** part/#asterisk cvance (46c48bac@gateway/web/freenode/ip.70.196.139.172) |
| 18:19.04 | Penguin | newtonr: Best policy is to assume no one knows anything until they prove otherwise. That keeps a level playing field. |
| 18:19.44 | *** join/#asterisk jkroon (~jkroon@197.96.209.127) |
| 18:20.14 | Penguin | And I'm not saying any specific person does or does not know... |
| 18:20.20 | Penguin | I'm saying I assume they do not know. |
| 18:20.55 | newtonr | Penguin, I agree |
| 18:21.14 | Penguin | If someone made the assumption that I DO know a certain thing, they would probably be wrong. I don't know much about anything, really. |
| 18:21.25 | jkroon | hi guys, I've got a VERY interesting scenario where a provider is sending me a re-INVITE with a=inactive, and then at a later stage sdp inside of an ACK, which I think is wrong, and won't cause asterisk to resume sending audio... ? |
| 18:21.55 | jkroon | broadsoft's app server should send the updated SDP in a re-INVITE, which asterisk will (should) then ACK ? |
| 18:22.37 | Penguin | One thing I do know, though, is that I am about to go to the microwave oven and pull out some type of monterey chicken dish I just cooked. |
| 18:24.20 | *** join/#asterisk slav3_kitten (~beepboop@unaffiliated/slav3-kitten/x-0866809) |
| 18:25.06 | newtonr | Penguin, I assume you don't know anything about cooking, so I hope it isn't awful. ;) |
| 18:25.32 | newtonr | Now I'm hungry |
| 18:25.39 | Penguin | Ah, I miss spoke. |
| 18:25.56 | Penguin | Stouffer's did the cooking. I just followed their instructions to make it hot. |
| 18:27.38 | Penguin | I will admit I didn't get it exactly perfect. Their instructions were based on an 1100W microwave, but mine most likely outputs some other power. |
| 18:27.47 | Penguin | inputs? |
| 18:27.58 | Penguin | Is that an input rating or an output rating? Who knows?! |
| 18:29.52 | newtonr | Stouffer's does make things easy. I often screw stuff up because of the power difference between the microwave at the office and mine at home. Always frustrating. |
| 18:31.14 | blitzrage | fwp |
| 18:32.04 | Penguin | Mine has always been over-powered compared to the estimated times on products. But I'm sure mine has a lower rating than their 1100W baseline. Maybe they are accounting for far too much inefficiency and loss; maybe my Panasonic does a better job than they predicted. |
| 18:40.13 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 18:40.37 | *** join/#asterisk starks (~eappleman@69.74.61.3) |
| 18:41.46 | starks | [TK]D-Fender, hi we spoke the other day. quick question to you and others... can i use cisco call manager as a sip provisioning tool? that is, configure sip on the cm and then push that configuration with the asterisk pbx proxy url? |
| 18:42.17 | [TK]D-Fender | I don't know CM. This is the wrong place for that info.... |
| 18:42.36 | starks | is there any gui whatsoever? |
| 18:42.47 | starks | or am i stuck troubleshooting scripts and ssh? |
| 18:43.01 | [TK]D-Fender | there are 3rd party bolt on ones like FreePBX |
| 18:43.03 | starks | i can't even get the phone to accept sip routing or sip options |
| 18:43.49 | starks | so... install freepbx, provision the phones, and then put the phones on the digium switchvox? |
| 18:44.02 | [TK]D-Fender | No... |
| 18:44.12 | [TK]D-Fender | FreePBX has nothing to do with Switchvox |
| 18:44.30 | [TK]D-Fender | Go look what Switchvox offers to make your life easy for them. |
| 18:44.36 | [TK]D-Fender | Other than that, it's on you |
| 18:45.01 | starks | it offers nothing in the way of dealing with cisco phones |
| 18:45.06 | starks | you fly blind |
| 18:45.17 | [TK]D-Fender | then it's all on you |
| 18:45.24 | starks | are you saying that freepbx does a better job? |
| 18:45.30 | starks | and handles this use case? |
| 18:45.34 | [TK]D-Fender | It DOES a job |
| 18:45.53 | [TK]D-Fender | Switchvox = no provisioning for Cisco. FreePBX's EPM can |
| 18:46.01 | [TK]D-Fender | (most cases) |
| 18:46.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.204) |
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| 19:10.11 | ibercom | which version of asterisk must consume less CPU ? 1.8 or 11, same configuration, g711, realtime, ... |
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| 19:13.30 | *** join/#asterisk ghost75 (~quassel@dslb-088-064-223-159.pools.arcor-ip.net) |
| 19:14.18 | ghost75 | if originate over ami is done and the remote peer is busy, will the return code be success or failure? |
| 19:16.05 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 19:16.46 | mjordan | Originate succeeds when the channel you originated is Answered. Otherwise, it fails. So the answer is: it depends on your dialplan. |
| 19:18.08 | Qwell | over AMI, it further depends on whether it's async or not |
| 19:23.26 | mjordan | Qwell: truth |
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| 19:27.11 | ghost75 | what is async? |
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| 19:29.37 | *** join/#asterisk Chotaire (~chotaire@host-089-207-249-134.vipri.net) |
| 19:29.49 | elenril | hi |
| 19:29.56 | *** join/#asterisk happy-dude (~Adium@darwin-mbp2012-sxc.wireless.rit.edu) |
| 19:29.58 | elenril | i'm having trouble with asterisk 11.7 and nat |
| 19:30.09 | *** part/#asterisk happy-dude (~Adium@darwin-mbp2012-sxc.wireless.rit.edu) |
| 19:30.17 | elenril | both the server and the client is behind nat |
| 19:30.20 | mjordan | ghost75: You can either originate a channel and block until the originated channel answers. Or you can originate and immediately return. |
| 19:30.36 | ghost75 | ok |
| 19:31.01 | elenril | i set localnet and externaddr, but asterisk is still writing the internal natted address in sdp |
| 19:32.36 | elenril | tried looking at the code and i don't quite see how is get_our_media_address() supposed to work |
| 19:33.09 | [TK]D-Fender | elenril: pastebin configs and complete call debug. |
| 19:33.11 | [TK]D-Fender | ~pb |
| 19:33.11 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:33.12 | [TK]D-Fender | ^^^^^ |
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| 19:48.59 | elenril | sec, on it |
| 19:50.04 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 19:50.59 | navaismo | is anyone well versed on debian that can give a hand... Seems like i killed my debian asterisk server |
| 19:51.34 | elenril | [TK]D-Fender: ok, here's my stripped down sip.conf http://pastebin.com/jQgT7mPe |
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| 19:58.20 | elenril | [TK]D-Fender: http://pastebin.com/nY7FHTgL and here is the call log |
| 20:00.04 | ChannelZ-Wk | navaismo: in what fashion? |
| 20:00.36 | navaismo | apt-get unusable, cant enter in the asterisk cli, dependencies errors etc etc |
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| 20:02.39 | ChannelZ-Wk | hmm sounds a bit more catastrophic. Did you do an upgrade or something and it blew up? |
| 20:06.23 | navaismo | trying to install openvpn result in an upgrade then in a dpkg --configure -a then in apt-get install -f and now im screwed |
| 20:07.03 | navaismo | ChannelZ-Wk, here some background-->http://pastebin.com/HDJsESHm |
| 20:10.05 | [TK]D-Fender | elenril: I don't have the config part I requested |
| 20:12.13 | elenril | [TK]D-Fender: which other files do you need? |
| 20:15.04 | [TK]D-Fender | elenril: I missed a PB, sorry. You're missing "nat=yes" under [general] |
| 20:15.44 | elenril | tried that too, doesn't solve the problem |
| 20:15.47 | elenril | let me pastebin the logs |
| 20:18.57 | navaismo | ChannelZ-Wk, this is what i got when trying to connect to asterisk: rasterisk: /usr/lib/libz.so.1: version `ZLIB_1.2.3.3' not found (required by /usr/lib/i386-linux-gnu/libxml2.so.2) |
| 20:19.20 | elenril | [TK]D-Fender: http://pastebin.com/pAZf2sjR |
| 20:19.25 | elenril | this is with nat=yes |
| 20:19.39 | elenril | as you can see, it still puts 192.168.0.6 in the sdp transmitted to galaxy |
| 20:20.33 | Qwell | navaismo: Did you build libxml2, or did it come from a package? |
| 20:21.37 | [TK]D-Fender | elenril: "sip show settings |
| 20:24.01 | navaismo | Qwell, asterisk was working(in fact is still running) but after the dpkg --configure -a i cant enter in the cli |
| 20:26.32 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.204) |
| 20:27.19 | elenril | [TK]D-Fender: http://pastebin.com/YYM4ME4V |
| 20:27.47 | [TK]D-Fender | <PROTECTED> |
| 20:27.50 | [TK]D-Fender | missing the port |
| 20:28.05 | pabelanger | no, that is a bug I think |
| 20:28.09 | pabelanger | I get the same, IIRC |
| 20:28.20 | [TK]D-Fender | <PROTECTED> |
| 20:28.33 | elenril | do i have to set the port? |
| 20:28.43 | elenril | it's listening on both udp and tls |
| 20:28.46 | [TK]D-Fender | apparently |
| 20:29.45 | elenril | no change after putting :5061 there |
| 20:30.07 | [TK]D-Fender | :/ |
| 20:30.12 | [TK]D-Fender | not sure at this point... |
| 20:30.18 | [TK]D-Fender | pb the updated configs... |
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| 20:41.47 | elenril | [TK]D-Fender: http://pastebin.com/gpb3SCf8 |
| 20:48.40 | navaismo | ?die |
| 20:48.46 | navaismo | ~die |
| 20:48.46 | infobot | ACTION takes two shots to the head and crumples to the ground, lifeless. |
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| 21:06.29 | [TK]D-Fender | elenril: Yup, at a loss for now... |
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| 22:31.17 | *** join/#asterisk devlap (~teonmea@71.177.68.116) |
| 22:32.36 | devlap | So I want to setup a full simulated PBX for one of our new office locations. I have a server and some IP phones to play with. Right now we have a PRI and we are using 8 lines on it for public phone numbers. I wanna get something SIP related, so I don't need an asterisk compatible PRI card, and that is super cheap for testing. I don't need to get all 8 lines tested really. |
| 22:32.53 | devlap | I was looking at www.didforsale.com |
| 22:33.00 | devlap | What kind of service would I need to play with this? |
| 22:36.33 | WIMPy | If you don't want hardware, you need what we call an ITSP. |
| 22:36.38 | WIMPy | ~itsp |
| 22:36.38 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 22:37.32 | devlap | Well I have an empty server. |
| 22:37.39 | devlap | And I have a LAN for it, and some ip phones. |
| 22:38.09 | devlap | Just the PRI card, I don't. I was told SIP would work. |
| 22:38.34 | devlap | So didforsale is probably an itsp |
| 22:38.38 | devlap | but idk what I need. |
| 22:38.38 | WIMPy | For some definition of "work". |
| 22:39.01 | devlap | No, actually, a recommended consulting company said thats what we would want. |
| 22:39.12 | WIMPy | But many people rely on such services. So it's definitely possible to use them. |
| 22:39.31 | WIMPy | What exactely? |
| 22:39.40 | devlap | Said PRI is basically additional complexity and it would work just fine. |
| 22:40.09 | WIMPy | PRIs are much easier. |
| 22:40.12 | devlap | Getting some kind of SIP service from my telco basically through verizon. |
| 22:40.21 | devlap | And expensive. |
| 22:40.29 | WIMPy | But it depends on _your_ needs, wht the right way can be. |
| 22:40.32 | devlap | its 580$/m through verizon for a pri. |
| 22:40.38 | WIMPy | Depends on your location. |
| 22:40.43 | devlap | That is my location. |
| 22:41.03 | WIMPy | Here PRIs are much cheaper than the same amount of IP bandwidth. |
| 22:41.28 | devlap | We have 150/90 fibre for less than the cost of that PRI :) |
| 22:41.29 | WIMPy | Try some ITSPs. |
| 22:41.48 | devlap | Like I said, I was looking at one, I don't know what service I need, and they list like 10 different things all semi related. |
| 22:42.14 | devlap | What does a sip trunk get me? |
| 22:42.20 | WIMPy | Everyone wants to sell you stuff you sdon't need :-) |
| 22:42.52 | WIMPy | There is no such thing as a sip trunk. And waht a sip account gives you can differ quite a lot. |
| 22:43.01 | devlap | http://www.didforsale.com/index.php/products/sip-trunking |
| 22:43.03 | coppice | lots of people want to buy stuff they don't need :-) |
| 22:43.18 | WIMPy | coppice: That's when it worked. |
| 22:43.55 | devlap | I want 2+ lines in with roll over to another line if the first is busy, and 2 lines out minimum for testing. |
| 22:44.24 | WIMPy | Often you will have one account per phone number, but you could have a single account with multiple numbers. And the may or may not be a limit on the nuber of simultaneous channels. |
| 22:44.46 | devlap | Ok so they price it per "SIP Trunk" and # of DIDs. |
| 22:44.57 | devlap | DID's being a number, the trunk being? |
| 22:45.10 | WIMPy | There's no lines. And that concept didn't make much sense with PRIs, wither. |
| 22:45.54 | WIMPy | Usually one account. But as a sip trunk is nonsense technically, it could mean anything. |
| 22:45.59 | devlap | So what does make sense, because I am trying to figure out what what I want is or is called, and you just keep saying I'm wrong. I know this. |
| 22:46.22 | devlap | Just trying to give info here to get something usable. |
| 22:46.28 | WIMPy | Maketing speak has no reliable terms. |
| 22:47.06 | WIMPy | You have to read the product descriptions to find out what it is they really offer. |
| 22:47.42 | devlap | And what do I want that they offer? |
| 22:47.53 | WIMPy | What do you want? |
| 22:48.06 | devlap | I mentioned it above, again. |
| 22:48.09 | WIMPy | Numbers? Number of simultaneous calls? |
| 22:48.12 | WIMPy | Look for that. |
| 22:48.42 | devlap | Ok and who is cheap and offers number and simulataneous calls? |
| 22:48.52 | devlap | cheaper the better. its just a test |
| 22:49.00 | devlap | do want full features tho. |
| 22:49.25 | WIMPy | Cheapewr is usually not better. They are often cheaper for a reason. |
| 22:49.38 | devlap | Well if the service is spotty or goes down, I don't really care about that. |
| 22:49.59 | WIMPy | Ask google for VOIP price comparison sites. |
| 22:50.24 | devlap | I did. I am asking you if there is a cheap decent provider out there given that you probably have tried or heard of people using one that is good |
| 22:50.44 | WIMPy | None, that could be of use for you. |
| 22:51.01 | WIMPy | The bot offered you a list. |
| 22:51.25 | devlap | i saw. |
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| 23:06.32 | devlap | I want to do something where if my phone at work gets a call, and im not logged in, it calls my cell, and the callerid shows 1 followed by the callers phone number. What is that feature called for outbound sip channels where I can set the callerid display? |
| 23:07.53 | WIMPy | Either you need to be allowed to send any caller ID (in that case I guess that means including invalid ones), or you can send a diversion header. |
| 23:09.17 | devlap | Does the latter work with standard phones? |
| 23:09.29 | WIMPy | Yes |
| 23:09.42 | WIMPy | But I'm not sure how many ITSPs support it. |
| 23:10.40 | devlap | Diversion headers? |
| 23:10.51 | WIMPy | I have one where I can't set caller ID, but Diverting number, i.e. the exact opposite of what you need. |
| 23:14.07 | carrar | yes that is Diversion headers |
| 23:14.21 | carrar | and ITSP worth anything should support that |
| 23:14.24 | carrar | and=any |
| 23:14.31 | carrar | if not, dump them |
| 23:14.43 | carrar | Diversion headers with PAI headers |
| 23:27.34 | devlap | Alright for testing purposes I think I need 2 channels, one in one out simulataneous, and a handful of minutes for cheap, and header support I guess. |
| 23:28.38 | devlap | ima do a free trial from didforsale and go from there. |
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| 23:58.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.241) |