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| 00:15.15 | qakhan | i am trying to get if there is any current call in context before make 2nd call. if yes than what is the duration of current call |
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| 08:53.13 | SupaYoshi | anyone tips about registrering through NAT? |
| 08:56.11 | MaliutaLap | make sure you have externip and localnet set properly. ensure you SNAT the sip and rtp ports to the * server |
| 08:56.17 | SupaYoshi | I cant register through NAT even though having port 5060 open and RTP ports. 10000 - 20000 |
| 08:56.28 | MaliutaLap | don't do it if you can avoid it |
| 08:56.28 | SupaYoshi | My SIP client just won't start to register. |
| 08:56.31 | SupaYoshi | It times out. |
| 08:56.33 | MaliutaLap | ~sipnat |
| 08:56.33 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 08:57.21 | MaliutaLap | it's not about "open", it's about making sure the outgoing packets are DNAT'd properly - and the incoming ones are SNAT'd |
| 08:59.30 | SupaYoshi | Ah okay. |
| 08:59.51 | SupaYoshi | Well calls and everything work, I just cant register. :x |
| 08:59.55 | SupaYoshi | So i closed em again.. |
| 09:00.46 | SupaYoshi | first link doesnt work |
| 09:08.16 | SupaYoshi | Im having harder issues then them. |
| 09:08.22 | SupaYoshi | I have double NAT. |
| 09:08.35 | SupaYoshi | because my provider can't bridge my modem. |
| 09:08.47 | SupaYoshi | I have opened all ports on the modem/router of the provider, with DMZ. |
| 09:08.53 | SupaYoshi | And on the router I forwarded the ports. |
| 09:09.08 | SupaYoshi | However, still the registration doesn't connect from outside :/ |
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| 10:59.26 | jermy | I'm having fun issues with Caller ID |
| 11:00.08 | jermy | I assumed that the UK was always bad for getting the right settings, but it seems like the module in use (Openvox / opvxa24xx) is actually picking up the number fine |
| 11:00.13 | jermy | it just isn't sharing it with Asterisk |
| 11:04.17 | jermy | I'm not sure how to debug this other than put more debug output into the kernel module and try to see what's going on |
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| 13:49.28 | qakhan | i am trying to get if there is any current call in context before making 2nd call. if yes than what is the duration of current call. i check the balance in db before make a call and after call hangup minus the balance and update the table. which works fine. |
| 13:49.29 | qakhan | but if there is a company and it has multiple phone lines 1001, 1002, 1003 etc... how can i check current call duration if there is any before making 2nd call. |
| 13:49.29 | qakhan | e.g |
| 13:49.29 | qakhan | company has $1.00 balance. |
| 13:49.29 | qakhan | $0.05 per call/min rate so total mins 20 |
| 13:49.29 | qakhan | before make a call i check in db how much balance a company has. |
| 13:49.29 | qakhan | now if a phone line 1001 makes call then 1001 has 20 min to talk (now counter started) |
| 13:49.30 | qakhan | 1001 has talked 10 mins and call is still in process |
| 13:49.30 | qakhan | now another phone line 1002 make a call and again i check balance in db for balance, result is $1.00 |
| 13:49.31 | qakhan | 1002 has another 20mins to talk, but it is wrong because 1001 has already been used half of balance and keep using it. plus 1002 has 20min to talk and if next line 1003 dial it will also get 20 min and so on. until 1001 finish all balance and balance get update in table and a new line try to call. please suggest how can i resolve this issue. like phone companies do shared mins. is there any |
| 13:49.31 | qakhan | possibility in asterisk that we can setup shared mins among more the 1 phone lines |
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| 16:10.34 | SirLagz | Asterisk should only need UDP 5060 and RTP 10000-20000 forwarded right ? That's what I've found online... |
| 16:12.27 | file | for SIP, yes |
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| 16:12.47 | file | and if behind NAT the proper configuration in sip.conf so the signaling reflects the external address when talking to stuff off your local network |
| 16:15.16 | newtonr | SirLagz, https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT |
| 16:15.29 | SirLagz | file: ok cool. already got all the NAT stuff setup |
| 16:15.32 | SirLagz | newtonr: thanks |
| 16:15.44 | newtonr | that is not a comprehensive example, but i wrote it for one of the more common cases |
| 16:16.08 | SirLagz | Something is stuffing something up though, after a while I can call into my DID but cannot call out from my asterisk box |
| 16:16.37 | SirLagz | reloading it will fix it for a while but I must be missing something |
| 16:17.03 | newtonr | if the issue only occurs "after a while" be sure you don't have a SIP ALG or something in your firewall/router |
| 16:17.27 | SirLagz | SIP ALG...hmm I'll have a poke around. I'm using pfSense as my router if you're familiar with it |
| 16:17.37 | newtonr | when i was writing that tutorial i actually discovered that my home asus router had a built in SIP ALG that was goofing things up. I had to disable it |
| 16:17.42 | newtonr | i know pfSense, but i'm not familiar with it |
| 16:18.10 | file | pfSense can go a little wonky with the outbound UDP mapping |
| 16:18.27 | file | I haven't used it in awhile so can't remember the details |
| 16:18.56 | SirLagz | hmm ok. I've setup port forwarding though, so should that help things ? |
| 16:19.39 | newtonr | that is pretty much a requirement :D |
| 16:19.53 | newtonr | with a lot of ifs, ands, and buts |
| 16:19.57 | newtonr | but yeah |
| 16:20.04 | [TK]D-Fender | SirLagz: for pf you need to disable port randomization <- |
| 16:20.15 | SirLagz | googles |
| 16:20.35 | file | https://doc.pfsense.org/index.php/Static_Port |
| 16:21.19 | newtonr | SirLagz, when you get it figured out , feel free to write up a little tutorial and submit it as a comment somehwere on the wiki and I can try put it on a page :D |
| 16:21.43 | SirLagz | newtonr: will do |
| 16:26.17 | SirLagz | ok so I've setup that static port stuff. Guess I'll find out if it worked or not in a day or so |
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| 17:27.02 | igcewieling1 | Can anyone tell me what to look for in order to find out information on what sip timers need to be changed in order for asterisk to give up on an invite to an unreachable device and return from dial? |
| 17:29.32 | file | you want timer B |
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| 17:39.13 | qakhan | i am trying to get if there is any current call in context before making 2nd call. if yes than what is the duration of current call. i check the balance in db before make a call and after call hangup minus the balance and update the table. which works fine. |
| 17:39.13 | qakhan | but if there is a company and it has multiple phone lines 1001, 1002, 1003 etc... how can i check current call duration if there is any before making 2nd call. |
| 17:39.13 | qakhan | e.g |
| 17:39.13 | qakhan | company has $1.00 balance. |
| 17:39.13 | qakhan | $0.05 per call/min rate so total mins 20 |
| 17:39.13 | qakhan | before make a call i check in db how much balance a company has. |
| 17:39.14 | qakhan | now if a phone line 1001 makes call then 1001 has 20 min to talk (now counter started) |
| 17:39.14 | qakhan | 1001 has talked 10 mins and call is still in process |
| 17:39.15 | qakhan | now another phone line 1002 make a call and again i check balance in db for balance, result is $1.00 |
| 17:39.15 | qakhan | 1002 has another 20mins to talk, but it is wrong because 1001 has already been used half of balance and keep using it. plus 1002 has 20min to talk and if next line 1003 dial it will also get 20 min and so on. until 1001 finish all balance and balance get update in table and a new line try to call. please suggest how can i resolve this issue. like phone companies do shared mins. is there any |
| 17:39.16 | qakhan | possibility in asterisk that we can setup shared mins among more the 1 phone lines |
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| 17:41.30 | Penguin | Check if there is a call within the entire context? You'd have to add a line of dialplan to every single extension within that context which will track the context and then a second line to check it. |
| 17:42.11 | Penguin | Look at the GROUP function. |
| 17:44.14 | Penguin | You can track the calls using GROUP and GROUP_COUNT, in conjuction with a conditional statement (such as by using GotoIf). |
| 17:45.00 | [TK]D-Fender | Set a variable for the account in each channel and have an AMI script constantly polling active channels, adding up the durations and comparing against balances. |
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| 17:56.04 | file | realistically it's hard to get it right |
| 17:59.19 | qakhan | ok so how Sip trunk provider do billing |
| 18:02.12 | file | some reserve extra from the balance at the start of a call, some allow you to go over to an extent |
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| 18:13.35 | [TK]D-Fender | qakhan: Billing is easy, limiting ACTIVE CALLS is another thing |
| 18:14.04 | [TK]D-Fender | qakhan: And very few use Asterisk for this |
| 18:14.49 | file | simplistic billing is easy in comparison to trying to actively limit things |
| 18:21.05 | qakhan | so is it possible that 1 assigne a group to all 4 phone lines and check balance on group base |
| 18:22.29 | [TK]D-Fender | You can lok at running channels any way you want. This is your job as a programmer |
| 18:26.20 | qakhan | how can i get running chennal info |
| 18:26.47 | [TK]D-Fender | AMI <- |
| 18:26.47 | Penguin | Tracking the calls is the easy part. Billing is different. |
| 18:26.50 | [TK]D-Fender | Read the book. |
| 18:26.54 | [TK]D-Fender | You've been at this for years. |
| 18:27.12 | [TK]D-Fender | if you can't see the forest for the tree you're going to have a serious problem. |
| 18:28.50 | qakhan | ok |
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| 18:41.50 | igcewieling1 | file: thanks |
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| 20:43.38 | derekb | hi there. looking for a few tips on modifying a script in asterisk. im trying to make it so that if CALLERID(num) presented as a certain number, to change it to something else. I've tried a few things with ExecIF, and im getting invalid syntax errors left and right, Asterisk is 1.4.35. The script I'm trying to modify is: http://pastebin.com/ZbbyuyA4 |
| 20:43.56 | derekb | I'm not a coder by any means, but just looking for a point in the right direction. |
| 20:48.24 | WIMPy | the first point to look at is 'core show application ExecIf' or the wiki. |
| 20:49.00 | WIMPy | And for some examples, voip-info.org can be a good source, even if much there is rather out of date. |
| 20:50.46 | derekb | Yeah, the voip-info page was confusing me as in the USAGE section it had things capitalized and spaces in random places, but in the examples those were not present. |
| 20:50.49 | derekb | I'll keep tinkering. |
| 20:51.21 | WIMPy | Do you want to set some fixed mapping? |
| 20:52.25 | derekb | Basically I want it so if callerid(num) = 123, set callerid(num) to 456 |
| 20:52.44 | WIMPy | For calls from SIP phones? |
| 20:53.14 | derekb | no, calls from an offsite non-asterisk pbx to a peer on my * box |
| 20:53.35 | derekb | basically they pass clid in a shitty way, so i want to replace what they send the peer, modify it, then the peer routes to pstn |
| 20:53.42 | derekb | with the modified clid... if that makes sense :D |
| 20:54.07 | WIMPy | No generel way to fix it? |
| 20:54.29 | WIMPy | I.e. do you need different way for different IDs? |
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| 20:55.23 | derekb | its just this one customer that has a pbx that we don't program, and no one local does. they want to strip caller id for a certain site. so when my sip peer receives the callerid(num) from them, i want to set callerid(num) to "" |
| 20:55.53 | derekb | i basically just need to know where in my script to do the modification, and what syntax to use. |
| 20:57.13 | derekb | I Tried a few modifications, and get to this point: |
| 20:58.39 | derekb | <PROTECTED> |
| 20:58.39 | derekb | <PROTECTED> |
| 20:58.40 | derekb | [Jul 19 16:57:48] ERROR[11019]: app_exec.c:191 execif_exec: Invalid Syntax. |
| 20:58.40 | derekb | (using an ip phone to test, currently) |
| 20:59.13 | WIMPy | Who wrote that script? |
| 20:59.27 | derekb | I did, i was playing around |
| 20:59.37 | derekb | the base script that i posted via pastebin was NOT written by me. |
| 21:00.46 | WIMPy | Thougt so, but it seems to already contain some stuff for modifying caller IDs. |
| 21:01.21 | derekb | I noticed that too, it's based on certain things in the gui from what i can tell. |
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| 21:03.11 | [TK]D-Fender | derekb: you should not be using quotes in your set |
| 21:03.12 | gusto | WIMPy, hi, how are you? |
| 21:05.35 | WIMPy | Operating temperature exceeded. |
| 21:05.56 | derekb | [TK]D-Fender: hi, and i will try that! |
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| 21:09.02 | gusto | WIMPy, here it is warm too, but it is getting better |
| 21:15.19 | derekb | [TK]D-Fender: thank you once again. thats twice you've helped me :D |
| 21:15.50 | gusto | WIMPy, I just installed Scientific Linux 7 on my MacBook, it looks well ;-) however, some software is still missing, but my bluetooth headset works with pulse here (because it has older version of pulse and bluez) |
| 21:18.23 | derekb | Hm, looks like my sip trunk provider doesn't like it when I Strip out the caller id number :P |
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| 21:20.47 | derekb | is there any special that needs to be done to pass unknown or anonymous clid? |
| 21:21.05 | derekb | assuming it depends how my sip trunk provider is expecting the call to hit them. |
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| 21:43.38 | [TK]D-Fender | "core show function CALLERPRES" |
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| 22:06.15 | derekb | yep, been playing with setcallerpres |
| 22:06.40 | derekb | [TK]D-Fender, can I call 'setcallerpres' in an execif statement? |
| 22:07.56 | WIMPy | You can, but you should set CALLERID(num-pres) instead. |
| 22:09.35 | derekb | hmm |
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| 22:10.42 | derekb | i was using execif to set callerid(num) to "" if a certain value was seen. then was going to use another execif to do setcallerpres(prohib_not_screened) |
| 22:11.03 | derekb | but callerid(num-pres) would be better for this scenario? |
| 22:12.06 | WIMPy | It's the new way to do it. |
| 22:12.18 | derekb | is it supported in 1.4.35? |
| 22:12.27 | derekb | or is that your syntax too new for me? |
| 22:13.03 | WIMPy | Uh. I can't remember 1.4. |
| 22:13.13 | [TK]D-Fender | That is just decrepit... |
| 22:13.16 | derekb | haha fair enough |
| 22:14.01 | [TK]D-Fender | that branch and the 3 that followed it are all no longer supported |
| 22:14.01 | derekb | [TK]D-Fender: ...sorry? I'm not a coder. Just experimenting to manipulate something. |
| 22:14.33 | [TK]D-Fender | derekb: there is a 1.4 equivalent somewhere... |
| 22:14.49 | [TK]D-Fender | "core show functions" , "core show applications" |
| 22:14.52 | [TK]D-Fender | go look |
| 22:15.02 | derekb | thank you. |
| 22:15.11 | WIMPy | You probably need setcallerpres on 1.4, but I have no idea if it actually worked. |
| 22:16.01 | WIMPy | I do remember that it was rather random on 1.4, at least with DAHDI. |
| 22:20.37 | derekb | awesome, works |
| 22:20.38 | derekb | thanks |
| 22:24.17 | [TK]D-Fender | Let me guess... running Asterisk on a NAS box? |
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