| 00:00.03 | rgagnon | sounds to me to be at least T1/E1 devices. but if you're in a country with old technology, that would not help you |
| 00:00.41 | WIMPy | That IS (good) old technology. |
| 00:00.49 | pigeonflight | we have a cable provider that does fibre to the curb, my home line is VOIP for example |
| 00:01.01 | WIMPy | pigeonflight: Where? |
| 00:01.04 | pigeonflight | I know they do PBX solutions built on this type of thing |
| 00:01.05 | navaismo | anyone have a link to understand the pri debug messages? The GGG cant help right now |
| 00:01.14 | pigeonflight | WIMPy: I'm in Jamaica |
| 00:01.17 | WIMPy | GGG? |
| 00:01.20 | rgagnon | ?? you're going to have 20 people doing 3000 calls/day in your house? |
| 00:01.26 | pigeonflight | rgagnon: no |
| 00:01.36 | pigeonflight | rgagnon: the solution is for a client at their location |
| 00:01.41 | WIMPy | pigeonflight: I have no idea what the prices are like there. |
| 00:01.51 | pigeonflight | WIMPy: neither me :) |
| 00:01.54 | joobie | back |
| 00:01.57 | joobie | sorry someoen at the door |
| 00:01.58 | pigeonflight | WIMPy: totally research mode |
| 00:02.08 | joobie | WIMPy, this telco doesnt |
| 00:02.16 | joobie | .. cut the line that is if i shoot too many alarms |
| 00:02.21 | joobie | ive shot heaps before in the passt |
| 00:02.23 | rgagnon | Jamaica is on the NANP, but I don't know if they use T1's like the USA or not... but the cool thing is that the digium cards that connect to those are switchable for either |
| 00:02.27 | WIMPy | pigeonflight: You need to find someone knowing the local situation. Anything else will probably just be misleading. |
| 00:02.38 | navaismo | WIMPy, Good God Google |
| 00:03.03 | WIMPy | navaismo: Aye. Well, then just ask. |
| 00:03.32 | rgagnon | in general, you're going to need a computer to put asterisk in, probably a 4-port T1/E1 digium card, and a bunch of SIP phones |
| 00:03.37 | pigeonflight | I know we have T1s here, I don't know if the newer guys (the digital cable guys are offering that) |
| 00:03.39 | rgagnon | with the network to put the phones on |
| 00:03.57 | rgagnon | if you can get business class cable with enough bandwidth |
| 00:04.12 | WIMPy | rgagnon: Apart from what I said, that doesn't vfit the specification so far at all. |
| 00:04.15 | rgagnon | and they sell phone service, you won't need any cards. you can just have the incoming calls through the cable company |
| 00:04.27 | pigeonflight | what capacity would a 4 port T1 give us? |
| 00:04.32 | navaismo | WIMPy, im trying to figure out wht means those messages--->http://pastebin.com/rXdcmXpg |
| 00:04.37 | rgagnon | a t1 supports 24 phone lines |
| 00:04.41 | rgagnon | well... 23 |
| 00:04.54 | WIMPy | 92095 simultaneous calls. |
| 00:04.55 | rgagnon | 1 is reserved for signalling |
| 00:05.01 | WIMPy | 92-95 simultaneous calls. |
| 00:05.02 | pigeonflight | nice! |
| 00:05.09 | pigeonflight | :( |
| 00:05.12 | pigeonflight | :) |
| 00:05.20 | pigeonflight | that's funny... thought you said 9095 |
| 00:05.35 | pigeonflight | started thinking "that's a lot of peeps on hodl" |
| 00:05.46 | rgagnon | a quad t1 card would support that |
| 00:05.55 | WIMPy | In theory you don;t even need channels for calls on hold. |
| 00:05.59 | rgagnon | a single t1 circuit is 23 calls |
| 00:06.00 | joobie | hmm damn |
| 00:06.02 | joobie | still down |
| 00:06.19 | WIMPy | But I don;t see that being usable with Asterisk. |
| 00:06.43 | pigeonflight | rgagnon: is that the $2300 Digium Wildcard TE412P? |
| 00:07.00 | rgagnon | oh right.. i missed the thing about 4 port t1... I saw "what capacity is a t1" |
| 00:08.07 | WIMPy | navaismo: You've got an issue activating the lower layers. |
| 00:08.44 | WIMPy | And BTW: I like how it says (TEI assigned) although that's not the case. |
| 00:09.17 | WIMPy | navaismo: Is that a PRI? |
| 00:10.39 | navaismo | supossed to be |
| 00:10.44 | rgagnon | pigeon ... or the TE420BF (depends on the slot in the computer)... also adding hardware echo cancellation about doubles the price |
| 00:10.47 | navaismo | no data from telco, |
| 00:10.54 | pigeonflight | rgagnon: so my checklist might include a 4port t1 + Digium Wildcard TE412P? |
| 00:10.58 | navaismo | im just shooting in the dark with all possibles configurations |
| 00:11.00 | rgagnon | IE the TE410P is much cheaper |
| 00:11.18 | rgagnon | the slot type generally does not change the price |
| 00:11.23 | pigeonflight | rgagnon: how important is echo cancellation? |
| 00:11.26 | WIMPy | navaismo: Yes, you're trying to activate layer2, but I guess you don't reveive anything. |
| 00:12.10 | rgagnon | pigeon: if you have the budget, get it |
| 00:12.17 | WIMPy | navaismo: If you set intense debug you can easily see if there is any data comming in at all. But I don't think it is. |
| 00:12.31 | rgagnon | I've seen some systems work fine without it, but others sound like you're in a tube underwater |
| 00:12.31 | navaismo | i dont have layer2 |
| 00:12.38 | navaismo | so using default i guess |
| 00:12.41 | navaismo | leave_down |
| 00:12.55 | WIMPy | navaismo: ? |
| 00:13.28 | rgagnon | actually noticed I have to leave .... sorry I cant' finish the conversation, pigeon... MUst pick up wife in 20 minutes, or hear about it for the next 15 yrs |
| 00:13.44 | rgagnon | good luck |
| 00:13.47 | *** part/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com) |
| 00:14.22 | navaismo | WIMPy, I dont have the layer2 line in the chan_dahdi |
| 00:14.46 | pigeonflight | so having 90 something lines, means I can have more persons on hold, correct? |
| 00:14.58 | navaismo | WIMPy, so the help say if i dont define that it use leave_down config |
| 00:15.18 | navaismo | WIMPy, this is the pri intense debug ouput--->http://pastebin.com/9TdfjGEE |
| 00:15.46 | WIMPy | navaismo: Ah. We don;t care if you will be notified. The issue is you cant get it up. |
| 00:16.34 | navaismo | right |
| 00:16.46 | navaismo | 8| |
| 00:16.48 | WIMPy | navaismo: As I expected not a single packet received ("<"). |
| 00:17.03 | *** join/#asterisk nightrid3r (~michel@62.205.65.222) |
| 00:17.07 | WIMPy | The link is up on both ends? |
| 00:18.30 | navaismo | well dont know how to translate that, but i guess yes based on connecting to old pbx it work, we can call |
| 00:18.38 | navaismo | but not with asterisk |
| 00:19.00 | *** part/#asterisk ghost75 (~trechber@dslb-178-002-158-225.pools.arcor-ip.net) |
| 00:19.24 | navaismo | tried with span=1,1,0,ccs,hdb3,crc4 || span=1,1,0,ccs,hdb3 || span=1,1,0,ccs,ami || span=1,1,0,ccs,ami,crc4 |
| 00:19.29 | WIMPy | What do the LEDs say on both ends? |
| 00:19.55 | WIMPy | That was E1? |
| 00:20.17 | navaismo | asterisk side red |
| 00:20.58 | WIMPy | Ok, that's not going anywhere then. |
| 00:21.43 | joobie | got it working WIMPy |
| 00:21.47 | joobie | fuken annoying |
| 00:22.01 | joobie | saturday 10am.. hpoefully no one noticed |
| 00:22.16 | navaismo | :( |
| 00:23.13 | WIMPy | navaismo: Are you using the same cable or a normal straight one? Have you tried both? |
| 00:23.33 | pigeonflight | how much ram etc.. would I need with a t1 4port? |
| 00:23.33 | navaismo | normal cable |
| 00:24.14 | joobie | pigeonflight, 4 x t1? |
| 00:24.27 | WIMPy | pigeonflight: Next to none. You need RAm for whatever else you plan to do. |
| 00:24.46 | navaismo | thanks WIMPy am off now |
| 00:24.48 | pigeonflight | has a lot to learn about voip/asterisk |
| 00:25.01 | pigeonflight | that's crazy... my webserver uses more ram than that |
| 00:25.34 | WIMPy | pigeonflight: Maybe you should find out first if you want to use VOIP or phone lines. |
| 00:25.50 | navaismo | going to drink one beer, his birthday o/ <8) |
| 00:26.17 | pigeonflight | my language is still imprecise, I meant asterisk, I'll probably start with phone lines |
| 00:26.29 | pigeonflight | then think about how to redirect calls to voip etc... |
| 00:31.04 | pigeonflight | in terms of a solid UI for asterisk (meaning the receptionist can manage things). I'm currently looking at Zentyal and FreePBX |
| 00:31.48 | WIMPy | Cool. Never heard about Zentyal before. |
| 00:32.57 | pigeonflight | Zentyal looks more complete, I'm just worried about getting tangled in it |
| 00:33.17 | WIMPy | Don't worry. You will be. |
| 00:33.58 | WIMPy | "I chose not to choose life. I Chose to Asterisk." |
| 00:36.02 | joobie | pigeonflight, why dont you look for a windows pbx solution if u want a gui? |
| 00:36.22 | pigeonflight | joobie: I don't know if I can survive on windows |
| 00:36.29 | WIMPy | Or a real PBX. |
| 00:36.30 | pigeonflight | joobie: it's been so long |
| 00:37.10 | pigeonflight | WIMPy: I know this client is budget sensitive, but they do want a reliable solution |
| 00:37.51 | pigeonflight | the UI is really so that they can manage and monitor things in my absence |
| 00:37.53 | WIMPy | What kind f features do they need? |
| 00:38.29 | pigeonflight | receive and transfer calls, have calls on hold (preferable with music or messages) |
| 00:38.39 | pigeonflight | it's a customer care type scenario |
| 00:39.30 | pigeonflight | If it were in the US I'd probably just recommend some cloud based solution like ringcentral |
| 00:40.25 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:40.26 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:20.09 | *** join/#asterisk Rahail (~Rahail@67.214.121.181) |
| 01:20.33 | Rahail | Hi there i got question how can you use dual interfeed coming to your asteirsk box |
| 01:21.11 | Rahail | letsay i want iax2user 1 use feed A and iax2 user 2 use feed B |
| 01:21.17 | WIMPy | Define "interfeed". And some punctuation might might your question more readable. |
| 01:21.17 | Rahail | how can achive this any idea |
| 01:21.26 | Rahail | !interfeed |
| 01:21.36 | WIMPy | Multihomed? |
| 01:21.59 | Rahail | no we are setting in africa and net is not so stable |
| 01:22.09 | Rahail | so we need to use 2 different net feed |
| 01:22.22 | Rahail | and also restricted to 1 feed x amount of data |
| 01:22.29 | WIMPy | And what's the issue with that? |
| 01:22.46 | WIMPy | It all depends on your routing. |
| 01:23.05 | Rahail | i need way to find iax2 user 1 send registration with feed A |
| 01:23.15 | Rahail | and iax2 user 2 send register with feed b |
| 01:23.29 | WIMPy | Thell them to do so. |
| 01:23.29 | Rahail | i try it with dual wan route |
| 01:23.51 | Rahail | both goes under same feed |
| 01:24.08 | Rahail | so i thought maby if i install 2 nic in asterisk box and if i can achive this |
| 01:24.17 | igcewieling | Rahail: generally Asterisk does not support custom network routing very well. |
| 01:24.53 | Rahail | igcewieling is there any method you could recomend |
| 01:24.57 | igcewieling | That is really more of an operating system thing. If the operating system routes the packets correctly it will generally Just Work with Asterisk |
| 01:25.08 | WIMPy | It all depends on your routing. |
| 01:25.50 | Rahail | WIMPy I understand routing but how ? I did try it use with pfsense dual wan |
| 01:25.54 | Rahail | ditn work |
| 01:25.56 | igcewieling | Rahail: you don't NEED two NICs, but you DO need the IP addresses on the NIC to be on different subnets. Doing it that way makes it easier. |
| 01:26.23 | igcewieling | Also if you expect this to work you should fully understand IP routing |
| 01:26.27 | WIMPy | How is your Asterisk connected to the internet? |
| 01:26.45 | Rahail | WIMPy directly connect to modem |
| 01:26.52 | Rahail | cat5 |
| 01:27.00 | WIMPy | So it has two public IPs? |
| 01:27.19 | Rahail | they give private ip |
| 01:27.24 | Rahail | its nated network |
| 01:27.58 | [TK]D-Fender | * will not multi-home by peer like that |
| 01:28.01 | WIMPy | Then the NATting device needs to take care of it. |
| 01:28.19 | Rahail | I try few way |
| 01:28.40 | Rahail | like on iax.conf i did bindportadd 192.168.100.15:4569 |
| 01:28.54 | Rahail | add antoher line bindportadd 10.10.20.105:4568 |
| 01:29.02 | Rahail | then i created 2 register string with ports |
| 01:29.06 | Rahail | that didnt work to |
| 01:29.16 | WIMPy | You can't have multiple bindports. |
| 01:29.26 | WIMPy | Either one or all. |
| 01:29.49 | [TK]D-Fender | There is no such thing as multiple bindports |
| 01:29.53 | Rahail | i am like lost even pfsens routing didnt work and i cant figerout what i am doing wrong |
| 01:29.54 | [TK]D-Fender | * binds to one or all. |
| 01:29.59 | [TK]D-Fender | not multiple individual |
| 01:30.04 | [TK]D-Fender | And * will not SOURCE them by peer |
| 01:30.08 | [TK]D-Fender | that is not happening |
| 01:30.16 | [TK]D-Fender | This is basic kernel routing |
| 01:30.25 | Rahail | [TK]D-Fender then how can i achive that |
| 01:30.45 | [TK]D-Fender | * cannot do this . |
| 01:30.49 | WIMPy | Your firewall will have to do it for you. |
| 01:30.56 | [TK]D-Fender | Closest equivalent would be to run TWO instances |
| 01:31.17 | WIMPy | Your Asterisk server seems not to be involved at all by your description. |
| 01:32.23 | Rahail | so there is no way use asterisk multipath |
| 01:33.12 | WIMPy | Off course you can, but Asterisk won't help you. You have to do it by configuring your routing. |
| 01:33.24 | Rahail | in server |
| 01:33.33 | Rahail | same server where asterisk or put router between |
| 01:34.05 | WIMPy | Where the routing is happening. |
| 01:34.50 | Rahail | i want it to do it same system where asterisk isntall |
| 01:34.59 | Rahail | but dont have much knowledge this why i was using pfsense |
| 01:35.02 | Rahail | as there gui |
| 01:35.14 | Rahail | so that way I do not have to install 2 system |
| 01:35.51 | WIMPy | This is confusing. What does your setup look like? |
| 01:36.27 | WIMPy | But anyway. Without some solid routing knowledge that isn't going to work out. |
| 01:36.44 | Rahail | this how it is |
| 01:36.53 | Rahail | isp modem--->asterisk |
| 01:36.55 | Rahail | right now |
| 01:37.04 | Rahail | and ISP modem gives me private ip |
| 01:37.17 | WIMPy | So that's not a modem, but a router. |
| 01:37.31 | WIMPy | And if that's the router that's where the magix happens. |
| 01:37.35 | Rahail | i have no access to it |
| 01:37.47 | Rahail | isp will not give us access to it... |
| 01:39.31 | [TK]D-Fender | Rahail: * will not muli-home by peer. that idea is dead. There is no way for that. Run 2 *'s instances |
| 01:39.59 | Rahail | then i have to use virtule |
| 01:40.00 | Rahail | ? |
| 01:40.27 | [TK]D-Fender | Rahail: How is up to you |
| 01:40.35 | Rahail | any other way around |
| 01:40.45 | [TK]D-Fender | no |
| 01:41.30 | Rahail | on virtulebox i just tell which vm to use which nic |
| 01:41.32 | Rahail | ? |
| 01:41.51 | [TK]D-Fender | Rahail: Each VM is its own machine. |
| 01:41.55 | WIMPy | What do you connect the 2nd NIC to? |
| 01:42.11 | Rahail | 2nd isp modem |
| 01:42.16 | WIMPy | Cool. I just managed to crash th remote console, but not the server process. |
| 01:42.52 | WIMPy | It's impossible for us to help you if you don't tell us what you have. |
| 01:43.00 | Rahail | ha |
| 01:43.03 | WIMPy | Again: How is your Asterisk connected to the internet. |
| 01:43.14 | Rahail | via ispA |
| 01:43.27 | WIMPy | EXACTELY |
| 01:46.27 | Rahail | tjhe |
| 01:46.29 | Rahail | then |
| 01:49.01 | [TK]D-Fender | [21:23]Rahaili try it with dual wan route |
| 01:49.03 | [TK]D-Fender | [21:23]Rahailboth goes under same feed |
| 01:49.19 | [TK]D-Fender | Rahail: your description of how your network is set up is VERY bad |
| 01:49.51 | Rahail | IspA/B-->pfsense with dual nic --->asterisk |
| 01:50.02 | [TK]D-Fender | rrahWe have no idea what is getting IP's. What modems are involved. What the real subnets are. What anything is even plugged into. |
| 01:50.47 | [TK]D-Fender | Rahail: Run 2 *'s. Have pfsense NAT them out their respective connections an NOT load-balance them |
| 01:51.50 | Rahail | I wanted to skip pfsense |
| 01:52.06 | Rahail | so we could save power and pc |
| 01:52.11 | [TK]D-Fender | Then set up each connection DIRECTLY on each * box |
| 01:52.46 | WIMPy | What's that pfsense doing anyway? |
| 01:52.57 | Rahail | at this point nothing |
| 01:53.01 | Rahail | i toke it out |
| 01:53.11 | Rahail | and connect the modem directly to asterisk box |
| 01:55.19 | eureka^ | whoa, Rahail is connecting from an address in the same DC as my asterisk gear |
| 01:55.20 | eureka^ | funky |
| 01:55.32 | Rahail | lol |
| 02:05.12 | *** join/#asterisk Caplain (~shayne@67.214.115.93) |
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| 05:44.52 | *** join/#asterisk linocisco (~linocisco@203.81.72.83) |
| 05:45.01 | linocisco | hi all |
| 05:47.09 | linocisco | there is something I want to know calling terisk server. Whenever I called to a PSTN nummber with asterisk server, I heard 2 dial rings and then heard it is redirecting to asterisk as series of rings. Thoese first sequeence and 2nd going to asterisk rings are not the same. I would like to known why |
| 05:47.29 | linocisco | there is something I want to know calling asterisk server. Whenever I called to a PSTN nummber with asterisk server, I heard 2 dial rings and then heard it is redirecting to asterisk as series of rings. Thoese first sequeence and 2nd going to asterisk rings are not the same. I would like to known why |
| 05:50.36 | igcewieling | linocisco: remove the "r" option to your Dial statement |
| 05:53.24 | *** join/#asterisk voxter_ (~voxter@d23-16-68-172.bchsia.telus.net) |
| 05:54.08 | themrrobert | Been working at this problem for weeks, Can anyone get anything from this? http://66.162.158.195/0Problem.pdf |
| 05:57.52 | *** join/#asterisk linocisco (~linocisco@203.81.72.82) |
| 05:58.25 | linocisco | igcewieling, sorry my internet was down. remove "r" option in which file and which command line? |
| 06:01.20 | *** join/#asterisk iprouteth0 (~james@unaffiliated/iprouteth0) |
| 06:01.41 | iprouteth0 | who is using SIP with TLS? |
| 06:02.13 | iprouteth0 | I am generating keys and certificates and I am wondering if the hostname/IP address option is required for client certificates |
| 06:02.39 | iprouteth0 | ./ast_tls_cert -m client -c /etc/asterisk/keys/ca.crt -k /etc/asterisk/keys/ca.key -C phone1.mycompany.com -O "My Super Company" -d /etc/asterisk/keys -o malcolm |
| 06:02.47 | iprouteth0 | the -C option |
| 06:04.41 | iprouteth0 | says it's just the common name so I'm wondering if it's a hostname does it ever get resolved or does it just need to be unique? |
| 06:08.59 | linocisco | igcewieling, sorry my internet was down. remove "r" option in which file and which command line? |
| 06:20.33 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.124) |
| 06:22.12 | *** join/#asterisk linocisco (~linocisco@203.81.72.87) |
| 06:22.20 | linocisco | igcewieling, sorry my internet was down. remove "r" option in which file and which command line? |
| 06:22.54 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 06:25.12 | [TK]D-Fender | [01:50]igcewielinglinocisco: remove the "r" option to your Dial statement |
| 06:25.16 | [TK]D-Fender | linocisco: ^^^^^ |
| 06:28.03 | *** part/#asterisk zman099 (~sdfsdfwer@ip68-5-15-41.oc.oc.cox.net) |
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| 06:42.17 | *** join/#asterisk linocisco (~linocisco@203.81.72.82) |
| 06:42.41 | linocisco | [TK]D-Fender, there are many lines under extensions.conf. I dont know which one |
| 06:46.41 | linocisco | [TK]D-Fender, there are many lines under extensions.conf. I dont know which one |
| 06:48.27 | [TK]D-Fender | linocisco: what part of "dial statement" is unclear? |
| 06:49.21 | linocisco | [TK]D-Fender, I dont know where to locate "dial statement" u mean |
| 06:49.36 | [TK]D-Fender | linocisco: Look. At. Your. DIAL. Commands for use of the "r" OPTION. |
| 06:50.09 | linocisco | [TK]D-Fender, what are my Dial commands? where to find? |
| 06:50.17 | [TK]D-Fender | EXTENSIONS.CONF |
| 06:51.27 | linocisco | [TK]D-Fender, yes. I am finding in extensions.conf. but found many lines. dont know which line to edit |
| 06:51.40 | themrrobert | wow TK surprised you haven't exploded yet |
| 06:53.08 | themrrobert | will look like exten => NXXNXXXXXX, 2, Dial(${EXTEN},Ttr); |
| 06:53.15 | linocisco | themrrobert, u all r expert. i m beginner |
| 06:53.37 | [TK]D-Fender | linocisco: You don't seem to have any clue about your own configuration. |
| 06:53.45 | [TK]D-Fender | linocisco: look where you used DIAL <- |
| 06:53.55 | [TK]D-Fender | ou know.. the thing that DIALS devices in your dialplan. |
| 06:59.44 | linocisco | themrrobert, I found that lines. wht do I do? |
| 07:00.04 | [TK]D-Fender | linocisco: Stop using the "r" option. |
| 07:08.06 | *** join/#asterisk linocisco (~linocisco@203.81.72.82) |
| 07:08.25 | linocisco | [TK]D-Fender, I found [trunktollfree] |
| 07:08.26 | linocisco | exten =>_91800NXXXXXX,1, DIal (${GLOBAL(TRUNK)}/S{EXTEN:${(GLOBAL(TRUNKMSD)}}) |
| 07:08.40 | linocisco | [TK]D-Fender, I found no "r" inside those lines |
| 07:08.54 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 07:54.39 | iprouteth0 | urgh.....trying to get SRTP to work with csipsimple |
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| 08:29.05 | sjs205 | I've found this link for details of the sippeers table: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure Can anyone point me to an explination of each of the fields? |
| 08:35.15 | ChannelZ | I believe they aught to pretty much mirror the config directives in sip.conf of which there are explanations |
| 08:36.50 | sjs205 | ChannelZ, Thank you... it is early and I've only had one coffee thus far :/ |
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| 09:04.49 | iprouteth0 | what should I be looking at for troubleshooting SIP TLS and SRTP |
| 09:05.42 | iprouteth0 | SIP w/ TLS registers. When I make a call attempt I get a 503 connection refused with the SRTP |
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| 09:16.55 | sjs205 | Anyone able to tell me what the lastms field is for in the asterisk database? |
| 09:17.03 | sjs205 | *sip.conf |
| 09:18.03 | ChannelZ | probably the last qualify response time for the peer |
| 09:20.20 | sjs205 | Thank you again ChannelZ, do you know whether this can be the same value for all users? |
| 09:21.15 | ChannelZ | Well assuming it is the qualify response time (like what you see in 'sip show peers') it's not something you set, it's something that gets set. It's a status value not a config option |
| 09:21.35 | ChannelZ | And I say "assuming" because I don't run Asterisk realtime so I'm just making a semi-educated guess |
| 09:22.21 | sjs205 | Well you educated guess is much better than my non-educated guess :) |
| 09:23.11 | sjs205 | ChannelZ, So would you say it is correct that this is only really required if a peer is set with host=dynamic? |
| 09:24.32 | ChannelZ | Well like I said it's not something YOU set. If you have peer qualification on, it basically sends an OPTIONS message to the device every so often like a ping to make sure it's still alive and accessable, and lastms would be the time it took the peer to respond the most recent time. |
| 09:26.19 | sjs205 | ChannelZ, Ahhh, I see... Sorry, let me explain what I am trying to do... integrate openSIPS and asterisk... my problem is that all of the database configurations that I have found do not allow host=dynamic to be set... So I am trying to amend the database to allow this to be possible... |
| 09:27.30 | sjs205 | As in the integration between the openSIPS databse and Asterisk... It would really help if I were a SIP expert... I'm not! ha :/ |
| 09:29.47 | ChannelZ | Sorry not sure I can help you there |
| 09:31.13 | sjs205 | Thank you ChannelZ, your help has been invaluable thus far :) |
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| 12:09.12 | profoX` | hi, I just updated Asterisk 1.6 to 1.8 on my Debian system and because audio seemed to have stopped working I tried to compile version 11 from source.. But the result is the same, I don't hear audio played by Playback and Background but actually calling works.. any idea on how to debug this? |
| 12:10.15 | profoX` | There is no error in the log, the playback seems to play/stream on the server but I just don't hear anything on my softphones (csipsimple on android and jitsi on linux) |
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| 12:19.38 | jkroon | hi guys, trying to figure out, given two peerings with two providers (either SIP or IAX/2 peers, doesn't matter), let's call them A and B. Now, I always first try on A, eg, Dial(SIP/A/${EXTEN}), now after that I can look at DIALSTATUS and HANGUPCAUSE. Somehow I want to make a decision whether or not passing the same call over SIP/B might actually work. Eg, is A down, or is the destination not reachab |
| 12:19.38 | jkroon | le? |
| 12:22.02 | WIMPy | And what is your question? |
| 12:22.52 | jkroon | if Dial(SIP/A/0123456789) returns, and was not answered, and didn't ring, ie, it failed, how do I know if doing Dial(SIP/B/0123456789) might succeed? |
| 12:23.10 | jkroon | for example, if 0123456789 is *busy* then obviously passing the call over B won't make a difference. |
| 12:23.25 | WIMPy | Exactely. |
| 12:23.27 | jkroon | or if the 0123456789 isn't allocated on the PSTN |
| 12:23.41 | jkroon | ok, so under what circumstances can I proceed to try the fall-back? |
| 12:24.17 | WIMPy | Hard to say. In the dyas of SIP the information you get back seems to be rather random. :-( |
| 12:24.49 | jkroon | haha, yea, the SIP code => CAUSECODE mappings seems to change with every version of * ... |
| 12:25.17 | WIMPy | Not only in Asterisk. The "professionals" have the same issues. |
| 12:25.18 | jkroon | well, pretty much I reckon for SIP anything in the 4XX range is final, 5XX and 6XX is not. |
| 12:25.51 | WIMPy | The easy thing is to rely on Asterisk and only retry on CHANUNAVAIL. |
| 12:26.29 | WIMPy | If you want to go in to detail, that probably involves testing out what your provider sends back to you under different circumstances. |
| 12:26.42 | jkroon | ok, so DIALSTATUS ends up being one of ANSWERED, BUSY, NO ANSWER, CONGESTION or CHANUNAVAIL where CHANUNAVAIL is *supposed* to indicate that we weren't able to signal the recipient? |
| 12:27.02 | WIMPy | And try it multiple times. Getting different results when trying multiple times seems to be normal as well :-( |
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| 12:27.09 | jkroon | WHAT?! |
| 12:27.18 | jkroon | that's dodge! |
| 12:28.05 | WIMPy | You can't rely on anything if anyone anywhere uses SIP. |
| 12:28.11 | jkroon | oh, and getting early media after a 183, having your client hang up and phone you about a number not existing and seeing "NO ANSWER" in your logs is confusing as hell. |
| 12:29.01 | WIMPy | Sounds interesting as well. |
| 12:29.41 | jkroon | POILs is such a PITA |
| 12:30.55 | WIMPy | POIL? |
| 12:34.46 | jkroon | Point-Of-Interconnect-Line |
| 12:35.17 | jkroon | if you've ever had to deal with a Telkom, Internet Solutions, Vodacom, MTN, Cell-C or 8-ta level of peering you'll know the politics. |
| 12:35.33 | jkroon | and SA law doesn't make it any easier. |
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| 13:21.26 | fling | Why am I having this error? > No application 'GoSub' for extension (hh, 00000, 1) |
| 13:22.08 | fling | Is not gosub accidentaly disabled by me after an update? |
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| 13:52.26 | jkroon | fling, "core show applications like Go" |
| 13:52.41 | jkroon | if it's not listed there ... then you need to load the appropriate module. |
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| 14:16.56 | jkroon | WIMPy, you're suggestion seems to mostly work, only picked up two HANGUPCAUSES which I don't like, and already deviced a way around that previously without realizing it. |
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| 15:35.30 | fling | jkroon: right, I do not see gosub |
| 15:35.35 | fling | what module am I missing? |
| 15:35.47 | fling | what should I load to get gosub working? |
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| 16:10.33 | igcewieling | fling: pastebin your /etc/asterisk/modules.conf |
| 16:11.54 | igcewieling | app_stack contains Gosub, Return, etc. |
| 16:13.01 | fling | igcewieling: /etc/asterisk/modules.conf > http://bpaste.net/show/91161/ |
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| 16:14.40 | igcewieling | good. now go back into make menuconfig and enable app_stack under Applications |
| 16:14.55 | igcewieling | then make and make install |
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| 16:25.39 | fling | igcewieling: gentoo here |
| 16:26.16 | igcewieling | fling: then uninstall Asterisk from whatever silly thing gento has and download and install asterisk from source. |
| 16:26.35 | igcewieling | or contact Gento and find out why app_stack is not compiled by default |
| 16:29.03 | fling | igcewieling: but it is |
| 16:29.05 | fling | <PROTECTED> |
| 16:29.11 | fling | loaded manually |
| 16:29.33 | igcewieling | "core show application gosub" does not work? |
| 16:29.50 | fling | -= Info about application 'Gosub' =- |
| 16:29.55 | fling | looks like now it works |
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| 16:30.16 | fling | do I need to add app_stack.so to preload in /etc/asterisk/modules.conf ? |
| 16:30.26 | igcewieling | no. |
| 16:32.06 | fling | so how to automagically `module load app_stack.so´ after each asterisk restart? :D |
| 16:34.22 | igcewieling | it does so automatically on every single asterisk install I've done since it existed. |
| 16:34.37 | igcewieling | which would be around 80 systems. |
| 16:35.16 | [TK]D-Fender | Go restart it now and see what happens |
| 16:35.25 | [TK]D-Fender | And show the actual load process and attempt |
| 16:35.46 | fling | mirror*CLI> core show application gosub ; Your application(s) is (are) not registered ; Command 'core show application gosub' failed. |
| 16:35.58 | fling | mirror*CLI> module load app_stack.so ; Loaded app_stack.so |
| 16:36.15 | [TK]D-Fender | Restart *. |
| 16:36.17 | [TK]D-Fender | SHOW US |
| 16:36.23 | fling | even if I add `preload => app_stack.so´ to /etc/asterisk/modules.conf |
| 16:36.25 | [TK]D-Fender | And dump the modules folder while you're at it |
| 16:37.32 | igcewieling | fling: preload is USELESS for almost all modules. |
| 16:37.46 | igcewieling | the exceptions being a few related to realtiime, odbc, and MoH |
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| 16:38.05 | igcewieling | [TK]D-Fender: he did not install from source, he installed from Gentoo's Portage. |
| 16:39.53 | fling | [TK]D-Fender: http://dpaste.com/1057244/ |
| 16:40.10 | fling | sip is loaded with preload |
| 16:40.12 | [TK]D-Fender | fling: Did not ask for a log |
| 16:40.15 | [TK]D-Fender | Restart * NOW |
| 16:40.17 | [TK]D-Fender | BY HAND |
| 16:40.25 | [TK]D-Fender | Dump the modules folder. |
| 16:40.28 | [TK]D-Fender | Show ALL in a pastebin |
| 16:40.42 | igcewieling | fling: and if you remove the reload of sip, it will still work. |
| 16:41.01 | igcewieling | if you remove the PRELOAD chan_sip it will still work |
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| 16:45.54 | fling | [TK]D-Fender: I don't understand :| sorry |
| 16:46.29 | fling | igcewieling: right, but [Apr 13 23:43:35] WARNING[15094] loader.c: Error loading module 'app_stack.so': /usr/lib64/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister |
| 16:47.11 | [TK]D-Fender | Well that's enough going in circles for me... |
| 16:47.13 | [TK]D-Fender | heads out |
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| 16:59.08 | igcewieling | fling: if you have no commercial or custom modules remove or rename /usr/lib/asterisk/modules and /usr/lib64/asterisk/modules and reinstall asterisk. |
| 16:59.21 | igcewieling | the error indicates you have module mismatches |
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| 17:01.06 | fling | igcewieling: ok, thanks |
| 17:02.13 | fling | igcewieling: `emerge -C asterisk´ and now I do not have /usr/lib/asterisk/modules and /usr/lib/asterisk |
| 17:02.31 | fling | igcewieling: so if I will reinstall it will appear again |
| 17:02.33 | igcewieling | I cannot help you with portage |
| 17:03.00 | fling | hmm hmm |
| 17:03.12 | igcewieling | if you insist on installing asterisk the hard way you are on your own |
| 17:03.28 | fling | All the modules are installed from the single version |
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| 17:18.50 | fling | [TK]D-Fender: igcewieling: solved using this solution > http://lists.digium.com/pipermail/asterisk-users/2012-September/274827.html |
| 17:21.45 | jeev | hm, when i call my DID from verizon, it gives me a ringback but it doesn't give me a ringback from tmobile |
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| 17:25.45 | fling | [TK]D-Fender: GoSub rocks! |
| 17:26.30 | jeev | ah something wong. |
| 17:29.51 | leifmadsen | GoSub() is pretty amazing |
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| 17:29.59 | leifmadsen | and not stack issues like Macro() |
| 17:32.31 | jeev | is it possible to have g729 between two asterisk servers, one at the datacenter, one locally.. then have it ulaw to the phones on lan ? |
| 17:33.17 | leifmadsen | yes |
| 17:33.28 | leifmadsen | you just set that up via the connections between asterisk boxes |
| 17:33.40 | leifmadsen | there are separate authentication paths between each of the end points |
| 17:33.45 | leifmadsen | that's how asterisk works, since it is a b2bua |
| 17:34.01 | leifmadsen | [asterisk_remote] |
| 17:34.05 | leifmadsen | disallow=all |
| 17:34.08 | leifmadsen | allow=g729 |
| 17:34.09 | leifmadsen | <PROTECTED> |
| 17:34.12 | leifmadsen | [my_device] |
| 17:34.14 | leifmadsen | disallow=all |
| 17:34.15 | leifmadsen | allow=ulaw |
| 17:35.11 | jeev | awesome |
| 17:35.19 | jeev | thought so but i was missing something. the g729 liense. |
| 17:35.20 | jeev | ;) |
| 17:35.31 | jeev | i'll continue with ulaw until i can get it |
| 17:36.04 | jeev | i can't believe how comfortable people are with using g729, it scares me about voice quality |
| 17:36.50 | igcewieling | Why? |
| 17:36.58 | igcewieling | g729 has great voice quality |
| 17:37.03 | igcewieling | try GSM sometime. |
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| 17:37.51 | jeev | hmm |
| 17:38.08 | jeev | igcewieling, it comes from ignorance, i never really tried g729. |
| 17:38.33 | igcewieling | the problem is g729 is the license 8-| |
| 17:39.09 | jeev | that's relatively cheap.. for such savings. |
| 17:39.17 | jeev | i mean when you have just 5 lines.. |
| 17:45.12 | jeev | anyway, i notice a lot of other places i call with tmobile, i dont get a ringback.. i'm dialing my DID with vzw and i'm getting a ringback, not with tmobile though |
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| 17:58.17 | mariusno | Hello, I am trying to figure out what "Condition 14" is. When I'm doing a Dial from motif |
| 17:59.01 | mariusno | context, i get that motif dont know what condition 14 is |
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| 18:18.21 | igcewieling | what exact version of Asterisk? |
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| 19:34.09 | vlad_starkov | Question: Asterisk 11 + sip realtime. Have a client behind NAT. The signaling works fine but RTP sends to "NATed" address of the client. This issue is not appears if I set static config for the client in sip.conf. Anyone knows how to deal with that? |
| 19:35.23 | vlad_starkov | In realtime and in static file nat=force_rport |
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| 20:02.03 | igcewieling | in sip.conf set localnet= and externip= |
| 20:02.15 | igcewieling | oh, worry. |
| 20:02.19 | igcewieling | sorry, that is not correct. |
| 20:02.28 | igcewieling | set directmedia=no either in sip.conf or in the peer entry. |
| 20:02.45 | igcewieling | and set nat=yes even though it generates a warning |
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| 20:08.18 | vlad_starkov | igcewieling: I'll try |
| 20:08.41 | igcewieling | vlad_starkov: there is a nat bug in recent asteisk 11 versions |
| 20:09.56 | vlad_starkov | igcewieling: nat=yes does not work |
| 20:10.09 | vlad_starkov | igcewieling: so is there a workaround? |
| 20:10.18 | igcewieling | nat=yes IS the workaround |
| 20:11.09 | igcewieling | could the NAT router the client is behind have SIP ALG or SPI enabled? |
| 20:11.16 | vlad_starkov | igcewieling: Using Asterisk 11.3.0. Set nat=yes in realtime database and no result. RTP still tries to send to client's local address |
| 20:11.37 | vlad_starkov | igcewieling: I'm afraid no |
| 20:11.49 | igcewieling | no. nat=yes in /etc/asterisk/sip.conf [general] section |
| 20:12.07 | igcewieling | SIP ALG and SPI will break Asterisk's NAT Fixup features |
| 20:13.40 | igcewieling | did you do a sip reload after changing the database info? |
| 20:13.43 | vlad_starkov | igcewieling: So in this case nat=yes will be default for all peers/friends. In case I have a peer with nonat I have to set in explicitly in peer's section? |
| 20:13.56 | vlad_starkov | igcewieling: I made asterisk restart :-) |
| 20:14.08 | igcewieling | nat=yes is harmless when used with clients not behind NAT |
| 20:14.15 | vlad_starkov | igcewieling: I'll try to put nat=yes in [feneral] |
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| 20:17.21 | vlad_starkov | igcewieling: nope. It still send rtp incorrectly |
| 20:17.55 | igcewieling | then remove realtime from the equation. |
| 20:19.42 | vlad_starkov | igcewieling: I can't. This node should have realtime due to be used by billing system. |
| 20:20.29 | igcewieling | I wish you the best of luck. |
| 20:20.38 | vlad_starkov | igcewieling: thanks!) |
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| 20:36.22 | vlad_starkov | What the difference between LTS and LTS certified versions of Asterisk? |
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| 20:43.19 | ChannelZ | Certified is if you have Digium phones |
| 20:44.27 | slav3_kitten | how are the digium phones? |
| 20:44.44 | Kobaz | pretty nice |
| 20:44.52 | Kobaz | i wish there was a cheaper basic model |
| 20:44.58 | Kobaz | like, a more basic, basic model |
| 20:45.05 | Kobaz | to hit the 90 dollar price point |
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| 20:48.25 | vlad_starkov | ChannelZ: thanks |
| 20:49.13 | vlad_starkov | 2nd attempt. Question: Anyone know how to deal with Asterisk 11 sip realtime + NAT issue? |
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| 20:53.26 | igcewieling | http://blogs.digium.com/2012/04/25/what-art-thou-oh-certified-asterisk-wherein-malcolm-does-his-best-to-explain-the-nifty-new-branch-of-asterisk/ |
| 20:54.12 | igcewieling | http://www.digium.com/en/products/asterisk/certified-asterisk |
| 20:54.21 | igcewieling | a basic google search would have answered your question |
| 20:55.19 | vlad_starkov | igcewieling: thanks) |
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| 21:05.43 | vlad_starkov | Oh. That was my mistake. The firewall was enabled. Sorry for inconvenience. |
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| 21:06.56 | Rahail | HI there letsay on my server I want to limit total calls for sip to recive 30 how can I do that however in additonal sip conf we can put for each user 20 40 like that 60 but we need 30 for hard limit |
| 21:06.58 | Rahail | how can we do that |
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| 21:14.45 | igcewieling | Rahail: see setvar= in sip.conf and "core show functions like group" |
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| 21:27.59 | Rahail | igcewieling |
| 21:28.03 | Rahail | sorry can u expalin me little more |
| 21:28.24 | igcewieling | read up on the GROUP functions on voip-info.org |
| 21:29.14 | igcewieling | You may want to learn Asteirsk better before trying to limit calls |
| 21:29.27 | Rahail | i know basic |
| 21:29.36 | Rahail | day by day learning new thing |
| 21:29.44 | Rahail | which is fun :) |
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| 21:31.03 | Rahail | i put under sip.conf calllimit that work however i made portal for regualr people when ever we add sip user it goes under additional sip conf however when we put the limit there that get increase |
| 21:31.24 | Rahail | even main sip.conf have 30 that no longer work if we have under aditional conf limi 10 40 50 per sip user |
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| 22:51.26 | iliterate | I'm having an issue with having multiple * servers behind a firewall since we have multiple sip trunks going to each server. The sip accounts register but the keepalives expire and then don't renegotiate because I can only forward the ports to one server. |
| 22:54.25 | nightrid3r | put an opensips/kamailio server between router and asterisk boxes and let that handle the routing for you |
| 22:57.09 | nightrid3r | s/router/firewall |
| 22:57.13 | iliterate | looking into it now. Thanks for your help! |
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| 22:58.35 | nightrid3r | keyword is sipproxy |
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| 23:30.56 | igcewieling | iliterate: is it a FIREWALL or is it NAT? |
| 23:31.59 | iliterate | NAT/Firewall/Gateway i'm using IPCop |
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| 23:34.04 | igcewieling | then you are set up to fail. |
| 23:34.19 | igcewieling | you have one of the few situations where NAT won't work correctly with SIP |
| 23:34.52 | igcewieling | put the internal asterisk boxes on different external IPs |
| 23:36.28 | igcewieling | You could also consolidate the two asterisk boxes into one asterisk box. |
| 23:36.44 | iliterate | that was going to be my last resort just throwing each one on a public ip. I was hoping to either configure my firewall to forward the correct port ranges that i specify in rtp.conf for each server but then the issue for 5060 and multiple internal ip's is an issue |
| 23:37.16 | iliterate | or maybe making one box for all sip trunk termination and use it as sort of a relay station that i can just register through to |
| 23:38.36 | iliterate | it just seems like redundant bandwidth if i throw each server on a public ip.. as it would be outgoing and then coming back in |
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| 23:59.18 | phix | hai gang! |