| 00:00.15 | carrar | a IPv6 version of phone calling? |
| 00:00.22 | carrar | 128 digit phone numbers |
| 00:01.18 | WIMPy | I don;t know what the time table is, but the international limits are about to be raised from 16 to 24 digits. |
| 00:01.49 | carrar | not looking forward to that |
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| 00:02.10 | carrar | What phone are you having issues with? |
| 00:02.30 | WIMPy | Eurit |
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| 00:03.20 | carrar | never heard of it |
| 00:07.21 | WIMPy | Ok, so now I know why we are supposed to change to the SIP shit. All the phones I tested now have a limit of 20 digits. |
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| 00:59.23 | WIMPy | #o2cantdo |
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| 04:18.13 | GreatSUN | hi all |
| 04:18.36 | GreatSUN | I am trying to get my hylafax iaxmodem stuff running again with asterisk |
| 04:18.57 | GreatSUN | something is surely wrong... registration seams to work (as peer) |
| 04:19.49 | GreatSUN | I added some outgoing route for extension 300 (the extension from iaxmodem) |
| 04:20.02 | GreatSUN | but I only get no dialtone |
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| 04:20.08 | GreatSUN | no local dialtone for sure |
| 04:22.30 | GreatSUN | [2013-12-06 05:22:15] NOTICE[30104]: chan_iax2.c:8761 update_registry: Restricting registration for peer '300' to 60 seconds (requested 300) |
| 04:22.37 | GreatSUN | is also coming all the time |
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| 06:11.29 | charkee | Hi guys, I have this weird problem with voicemail. It works as expected when dialled from the LAN/SIP. But when dialled over the PSTN/PRI no audio is heard. Regular calls over the PRI work though. Any ideas on how to approach this problem? |
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| 07:35.40 | nunne | anyone have any experience with Cisco UCM? i have a client that really needs to see it working together with asterisk (two different officies) and I have set up a trunk and gave him instructions according to a freepbx + cucm how-to i found. so bascially he can call my asterisk box. but I get busy/congestion trying to call his. So I have a feeling it's a problem with his dialplan... is there |
| 07:35.40 | nunne | a simple way to just re-direct all trafic from one sip-trunk to one ip-phone in CUCM? and how? any help is much appreciated. Since it's a deal breaker for them :P |
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| 08:16.26 | apwelsh | Anyone get Asterisk 11 to compile on OSX Mavericks with XCode 5? |
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| 08:27.15 | apwelsh | Anyone on here with OSX 10.9/Asterisk experience? |
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| 09:39.35 | kasanop | hi! is there an analog for nat=yes in chan_pjsip? |
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| 11:16.29 | gusto | what is chan_pjsip? |
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| 12:14.26 | file | kasanop, instead of overloading a single option to mean multiple things there are multiple options - force_rport, rewrite_contact, rtp_symmetric |
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| 12:18.36 | anonymouz666 | I am back |
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| 13:22.58 | anonymouz666 | Hi all. Just to let you know, great videos in asterisk youtube channel. |
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| 14:28.08 | sframe | Hi asterisk people. Where can I find information on testing DID number from pur provider? (they seem to stop working sometimes!) Any tutorials? 1K numbers that need to be tested daily. |
| 14:28.58 | [TK]D-Fender | testing means using |
| 14:31.18 | sframe | [TK]D-Fender: Sure. Is anyone aware of how to script dial numbers? I dont think I need an autodialer (was looking at wombat) but somehow run through list of numbers and noting which ones are *busy* or don't answer |
| 14:32.22 | dose_ | I'm using wombat is really good |
| 14:32.48 | dose_ | it has two free license for 2 channels |
| 14:33.03 | dose_ | it's a good start |
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| 14:33.20 | sframe | dose_: ok I'll into it some more |
| 14:33.41 | dose_ | they gave me 100 free channels for two months |
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| 14:34.44 | dose_ | the software is really easy to set up compare to astercc |
| 14:35.57 | dose_ | we are dialing around 10 000 numbers per day without problem, and I'm using AMD from Asterisk to detect human or machine |
| 14:40.14 | sframe | dose_: nice. I just need to know if it "rings" or not. do you think that will be a problem? |
| 14:42.17 | dose_ | what do you mean ? |
| 14:42.32 | dose_ | oh ok |
| 14:42.47 | dose_ | you just want to check if number exist |
| 14:42.48 | dose_ | ? |
| 14:46.35 | dose_ | in this case you should set in wombat feature "Answer timeout (ms.)" to few seconds to prevent dialed number to answer, when you campaign will finish to run through you number list in the report you'll have a list of all number with different state as "No answer, busy, error" so you can export this list and have a clean list of all number which ring |
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| 14:57.53 | saint_ | hi all - can someone give me some information about e911 ? I am trying to understand how it works. Is it something like rfc 5139 & 5491 that sends PIDF-LO to a PSAP , or does the PSAP have a database of VoIP phones / SIP trunks where the customer select to update automatically their location ? |
| 14:58.40 | JamKo | Why is the asterisk t.38 stack such a pile of garbage? I've extensively tested it 1.6, 1.8 and now 11.6. Every version has the same egregious behavior. |
| 14:58.42 | JamKo | It's a complete fail for asterisk as a whole, and is a result of unjustified stubbornness |
| 14:59.20 | WIMPy | You sound like you're new to Asterisk. |
| 15:00.20 | JamKo | Actually I'm not, I've been committed to making it work in all typical telco scenarios for years. The t.38 is a pile and not compatible with most commericial media gateways. |
| 15:00.51 | WIMPy | I'd be pretty happy if that was the only issue. |
| 15:01.09 | JamKo | It's f-ing bullshit. |
| 15:01.56 | JamKo | Any devs in here that work on this pile should get over your high horse stubborn BS and fix it. |
| 15:02.03 | WIMPy | If you can still get hold of them, always use real phone lines for fax. |
| 15:03.26 | JamKo | It could be made work. There are very specific behaviors that it has, that breaks most T.38 setups with the commercial equipment used by large carriers. |
| 15:03.43 | Chainsaw | JamKo: And your patch is in the bug tracker already? |
| 15:04.28 | JamKo | I hear what you're saying chainsaw, but I'm not a dev. I have the fix for the devs to implement if they care to resolve it. |
| 15:04.47 | carrar | Asterisk gives out the source for FFA? |
| 15:04.51 | carrar | err Digium |
| 15:05.08 | Chainsaw | carrar: I can tell lies as long as I put a question mark behind it? |
| 15:05.30 | carrar | heh |
| 15:06.22 | JamKo | requiring symetric latching to complete a t.38 setup is ridiculous. Asterisk sets up T.38 on a new port, and then sends from the old port. WTF is that? It's been that way all the way back to 1.6. |
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| 15:10.22 | file | define "on a new port" |
| 15:10.33 | JamKo | This has been submitted in previous bug reports. Fleming states it's a gray area in t.38 to have to send from the port negotiated in the reinvite until after the ACK.Really then why send from the old port at all? |
| 15:11.05 | file | are you referring to using the RTP port for T.38 as well? |
| 15:12.24 | JamKo | Asterisk reinvites to t.38 with a new port, by default 4000-4999. It then begins sending T.38 packets from the original rtp port negotiated in the first invite. |
| 15:13.01 | JamKo | Instead of sending from the new port it just negotiated. Granted the ACK is not received, but who cares? The other media gateway doesn't care about t.38 from the old port anyway. |
| 15:13.44 | file | I've never seen that... I don't even know how that could happen internally. What can happen is RTP packets in flight still go out. |
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| 15:14.30 | JamKo | I have about 60 captures showing this behavior in 3 different versions. |
| 15:14.53 | file | put one up somewhere then... |
| 15:15.13 | JamKo | To make things worst, Asterisk usually stops sending T.38 after the ACK is received, because it's expecting return packets from what it sent from the old ports. Hello??? |
| 15:15.17 | mjordan | JamKo: there is an issue for that, no one has worked on it. |
| 15:15.25 | mjordan | JamKo: if you'd like to help out with it that'd be great |
| 15:15.31 | file | mjordan, reallllllllllly? |
| 15:15.53 | mjordan | file: yeah, I vaguely recall that. There are some T.38 stacks that interoperate with Asterisk that will re-use the same port, and Asterisk isn't always a fan of it. |
| 15:15.53 | JamKo | mjordan: Yeah I'd really like to get it fixed. Thank you, |
| 15:16.18 | mjordan | goes to dig into JIRA. |
| 15:16.39 | coppice | some T.38 stacks will never do anything but reuse the RTP port |
| 15:17.01 | file | oh, I was thinking we were sending from the port |
| 15:17.48 | mjordan | https://issues.asterisk.org/jira/browse/ASTERISK-19013 |
| 15:18.01 | mjordan | that's the one I'm thinking of, at any rate |
| 15:18.22 | coppice | reusing the RTP port for UDPTL is a good strategy where NAT is involved. if the RTP port has punched its way through a firewall, you can be sure the UDPTL will also get through |
| 15:19.09 | *** join/#asterisk navaismo (~navaismo@189.241.125.73) |
| 15:19.19 | mjordan | JamKo: if you go to issues.asterisk.org and use the following search query "project=ASTERISK and component="Channels/chan_sip/T.38" and status != Closed and status != Complete" - and none of the issues look like what you're describing, feel free to open a new one. |
| 15:19.33 | mjordan | JamKo: However, keep in mind this is an open source project. Complaining about something will only get you so far. Code talks. |
| 15:19.56 | file | oh... yeah... I see |
| 15:20.00 | JamKo | ok. |
| 15:20.02 | file | coppice, quite |
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| 15:20.51 | JamKo | I have to run out for awhile but I'll get on it this afternoon. Thanks for the effort. |
| 15:21.51 | file | either have to reuse the same local port for UDPTL as RTP which would be impossible currently, or purposely introduce delay to allow the in-flight RTP packets to reach the other side and then send UDPTL |
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| 15:32.47 | darkunderlord | Can I use Realtime AND ODBC together for voicemail? |
| 15:33.03 | darkunderlord | I want Realtime, but it seems I can't use it for storage, which I want ODBC for |
| 15:35.12 | [TK]D-Fender | darkunderlord: IIRC you can only have 1 registered backend at a time for VM |
| 15:35.27 | [TK]D-Fender | darkunderlord: unlike CDR |
| 15:35.32 | darkunderlord | While I'm waiting, how many of you were at Astricon this year. This is Travis, maybe I saw you at the Fastest Dudes to the Dialtone contest? :) |
| 15:35.55 | darkunderlord | [TK]D-Fender, dang, any idea if they plan on adding storage to realtime at any point? |
| 15:36.25 | [TK]D-Fender | darkunderlord: I'm not a dev... can't say... |
| 15:37.13 | darkunderlord | can't understand why they wouldn't have implemented it. There must be some issue to using it with Realtime, otherwise it's a no brainer to make all old features available in the new |
| 15:41.40 | mjordan | darkunderlord: no plans to. The code that does the voicemail backends is archaic and scary and dragons lurk in there. |
| 15:41.55 | darkunderlord | lol |
| 15:42.06 | mjordan | darkunderlord: If you were at AstriCon (hi!), I'd say the longer term plan is to allow people to build their own voicemail systems outside of app_voicemail |
| 15:42.20 | file | we're getting closer bit by bit |
| 15:42.40 | darkunderlord | but the filesystem storage defeats the purpose of realtime and having clustering/failover |
| 15:42.53 | darkunderlord | I actually got second in the contest. :) was alot of fun |
| 15:43.26 | darkunderlord | guess I'll have to do some syncing etc |
| 15:43.32 | darkunderlord | on the file system level |
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| 15:46.10 | sframe | dose_: sorry had to go afk a sec, thanks for the tip! I'll try it out |
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| 16:35.11 | HolTech | Running a regular phone off a Sipura FXS port, not getting the CID number, the name comes across fine, tried setting the number arbitrarily in the dialplan, no success, I've tried using a different regular phone, same thing "Unknown Number" , any ideas? |
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| 16:35.59 | HolTech | i hsould not here that the CID # displays in the * log and console just fine |
| 16:36.03 | HolTech | *note |
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| 16:43.00 | darkunderlord | HolTech, that's probably a problem with your Sipura and the analog phone. I've honestly never tried doing that. |
| 16:44.10 | Chainsaw | It's always vital that you set up the FXS for the correct country signalling. |
| 16:44.38 | Chainsaw | As the analog telephone will have specific expectations. (Particularly if it is a DECT base station, it tends to need a full powercycle to clear any assumptions it has starting making about CID signalling) |
| 16:44.39 | HolTech | I am unfamilar with country signaling, is that the impedenance ? |
| 16:44.55 | Chainsaw | It's particularly how it transmits the caller ID, and when. |
| 16:45.05 | Penguin | There's a regional settings tab in the configuration interface. |
| 16:45.38 | Chainsaw | And if you should live in a region where CID is presented only after the first ring... |
| 16:45.43 | Chainsaw | Don't answer your test call too fast :) |
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| 16:48.28 | [TK]D-Fender | HolTech: Show us the call debug. |
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| 16:54.28 | HolTech | the sip debug ? |
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| 16:55.10 | Penguin | With core verbosity set to at least 3. |
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| 16:55.22 | Penguin | core set verbose 3, sip set debug on |
| 16:55.25 | Penguin | Then make your call. |
| 16:58.52 | HolTech | http://pastebin.com/2bLY92v2 |
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| 17:02.25 | [TK]D-Fender | HolTech: You should have known to have included SIP DEBUG so we can prove what's coming in and what's being sent... |
| 17:03.31 | HolTech | Apologies, i will go again |
| 17:03.55 | Penguin | <Penguin> core set verbose 3, sip set debug on <------------ |
| 17:07.28 | HolTech | http://pastebin.com/8wMtRr9s |
| 17:10.13 | HolTech | Call-ID: 2d425a892bfbd7ba4ee3d47f6d1d5ee7@192.168.1.138:5060 , line 274 , something tells me that could be wrong ? |
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| 17:10.58 | Penguin | The "From:" only shows the name. So that's why your phone only shows the name. |
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| 17:12.43 | HolTech | so what is between the < and > should be the #, any idea what I set wrong, the extension itself ? |
| 17:13.15 | Penguin | The call comes in with caller id on it. You shouldn't be setting it after that. If it is present, leave it alone. |
| 17:13.45 | HolTech | Right, this problem exists without any calls to "set" |
| 17:14.04 | Penguin | Good. |
| 17:14.16 | Penguin | I saw the number was in the PID. |
| 17:14.29 | Penguin | err... RPID. |
| 17:14.32 | HolTech | Could I be inproperly using fromuser field in sip.conf ? that is needed for the sip device to register i believe |
| 17:15.07 | Penguin | It is not needed for a device to register. That is very possibly the problem. |
| 17:15.24 | Penguin | Your device entry should not have a fromuser on it. |
| 17:16.48 | HolTech | That was it... took them out of sip.conf, did a sip reload and now all is well |
| 17:16.56 | [TK]D-Fender | fromuser forces the from when a call is sent mashing callerid |
| 17:17.06 | [TK]D-Fender | and it is not required i this case |
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| 17:17.58 | HolTech | I had assumed the fromuser and secret needed to match the Sipura's User ID and Password... always more to learn.. Much obliged, thanks guys ! |
| 17:18.12 | Penguin | For a device that registers, you should have the name (which is found in square brackets), secret, context, callerid if you prefer to force it, and host=dynamic. |
| 17:19.16 | Penguin | Don't add arbitrary settings. |
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| 17:25.33 | jonno11 | Hi guys. I'm getting audio on my phone, and laptop. However when my colleagues make any form of call, no audio is heard either way. This occurs on both iPhone and laptop. I've logged a working call, and a not working call. Would you be able to read through them and give some pointers of what's going wrong? https://gist.github.com/jonlambert/0a24d766a1f46d2c36a5 |
| 17:28.48 | Penguin | My guess is NAT is involved. |
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| 17:29.41 | jonno11 | Penguin: It is. We're all on the same network though |
| 17:29.49 | Penguin | Where's asterisk? |
| 17:30.04 | Shaka_Max | jonno11, the ip in the SDP is 10.0.231.151 while the ip in the sip siganlling is 194 so i guess nat issues |
| 17:30.07 | jonno11 | On a VPS, with a public IP |
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| 17:30.35 | jonno11 | Asterisk isn't behind a NAT |
| 17:30.45 | Penguin | Did you configure the asterisk peers for NAT? |
| 17:32.11 | jonno11 | Penguin: I think so - I have to admit I'm not 100% sure how I should be configuring it |
| 17:32.35 | Penguin | If devices are behind a NAT, you need to tell asterisk that they are behind NAT. |
| 17:32.38 | jonno11 | Penguin: I've updated the gist |
| 17:32.42 | jonno11 | With my sip.conf |
| 17:32.59 | Penguin | If you don't, asterisk cannot correctly deal with the private IP addresses found in the packets. |
| 17:33.05 | jonno11 | Penguin: There are 10/11 more |
| 17:33.10 | jonno11 | But identical |
| 17:33.23 | jonno11 | Penguin: Does it look ok? |
| 17:34.51 | Penguin | I wouldn't have chosen type=friend, but I do see nat=yes. Be very sure that your device is not also trying to trick asterisk with regard to NAT stuff. Turn off STUN/ICE/TURN or nat traversal in the phone. |
| 17:35.59 | Penguin | If you have asterisk correctly handling the NAT stuff and also try to get the phone to take care of it, that will cause problems. |
| 17:36.32 | HolTech | Can * really be used to spoof CID ? I would think the telco would over ride any CID you send out |
| 17:36.37 | jonno11 | Penguin: Ahh okay |
| 17:37.19 | jonno11 | Penguin: No go I'm afraid |
| 17:37.20 | Penguin | holtech: Asterisk will send whatever callerid info you tell it to send. What the phone company does with it after that is up to them. |
| 17:37.52 | jonno11 | Penguin: Still not working :/ |
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| 17:38.45 | HolTech | I'm not interested in doing anything mischievious, just curious because my telco ( Verizon ) charges $2 a month per line to set the caller id name, which apparently doesnt work anyways because I originally paid but people with Comcast land lines showed my calls as somebody else's name, most liekly the last owner of that number |
| 17:39.27 | Penguin | holtech: CallerID names are not sent to the remote party. |
| 17:40.18 | HolTech | So each carrier ( ie Comcast ) has a directory database they use then ? and that gets updated when they feel like it ? That was my impression |
| 17:40.31 | Penguin | holtech: Numbers are transmitted to the remote party, but the remote end has to look up the name in their database. If you call someone and they see another person's name associated with your phone number on the caller ID, that's their phone company's fault. |
| 17:41.08 | jonno11 | Shaka_Max: Any ideas how to fix it? |
| 17:41.23 | jonno11 | Thanks for your help guys |
| 17:41.36 | Penguin | If every company uses the same database and always updates it, everyone's name would display correctly on calls. |
| 17:42.16 | Penguin | Since there are various LIDBs out there, if you update one and the remote party's phone company users another, they'll either display a generic name or the wrong name. |
| 17:42.17 | jonno11 | ^ just realised that sounded sarcastic haha, didn't mean it that way |
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| 17:43.29 | Penguin | jonno11: How many phones do you have in this same location? How many work and how many do not work? |
| 17:43.56 | HolTech | jonno11: Try turning off iptables if you can, just to test... that was once my problem, I never did sort about exactly what ports needed for RTP, I thought 10000-20000 which is in the conf file, but still on my todo list |
| 17:44.46 | jonno11 | Penguin: My laptop + my iPhone are working. Two of my colleagues' phones, and their laptop is not. Tried multiple accounts and multiple sip clients |
| 17:45.15 | jonno11 | HolTech: I'll give that a try now |
| 17:45.39 | HolTech | and any firewall on the laptop, at least that way you can isolate the issue to the firewall and not * config |
| 17:50.38 | jonno11 | HolTech: I don't think there's a firewall on iPhone... but I've disabled iptables and that's a no go :/ |
| 17:52.17 | Shaka_Max | jonno11, Usually i configure the externhost/externip and the localnet even if asterisk is not behind nat just to ensure he know who are outside |
| 17:52.41 | jonno11 | Shaka_Max: Ok - what do I configure them to? |
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| 17:58.12 | Penguin | Set externaddr to your static IP address. |
| 18:00.59 | Penguin | I don't think it will help the problem, but what can it hurt? |
| 18:01.43 | Shaka_Max | its like the last glass of wine before leaving the party |
| 18:03.18 | jonno11 | Shaka_Max, Penguin: already done I'm afraid |
| 18:04.29 | Shaka_Max | jonno11, which version of asterisk? |
| 18:05.42 | jonno11 | Shaka_Max: 11.6.0 |
| 18:05.47 | jonno11 | Shaka_Max: Built from source |
| 18:07.28 | Shaka_Max | have you tried with the other options |
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| 18:07.38 | Shaka_Max | comedia |
| 18:07.46 | Shaka_Max | force_rport |
| 18:08.00 | Penguin | yes = comedia,force_rport |
| 18:08.21 | jonno11 | where's that? |
| 18:08.41 | Penguin | nat=yes is the same as nat=comedia,force_rport |
| 18:08.53 | Shaka_Max | thats why i didnt put that togheter |
| 18:09.06 | developerhealey | hello... i am looking for assistance with this command: In my extensions_custom.conf file I want to say if the User extension typed = 111 then i want to set it to 222 |
| 18:09.31 | [TK]D-Fender | developerhealey: that is not a standard asterisk config file |
| 18:09.43 | developerhealey | exten => s,n,Set(my_ext=${If($["${my_ext}" = "102"]?"270":${my_ext})}) |
| 18:10.01 | [TK]D-Fender | developerhealey: There is also no such thing as "set it to 222" |
| 18:10.09 | developerhealey | i am sorry, has nothing to do with the file, just the Asterisk commands syntax being used correctly |
| 18:10.13 | Shaka_Max | jonno11, ; nat = no ; Do no special NAT handling other than RFC3581 |
| 18:10.13 | Shaka_Max | ; nat = force_rport ; Pretend there was an rport parameter even if there wasn't |
| 18:10.13 | Shaka_Max | ; nat = comedia ; Send media to the port Asterisk received it from regardless |
| 18:10.13 | Shaka_Max | ; ; of where the SDP says to send it. |
| 18:10.13 | Shaka_Max | ; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default) |
| 18:10.14 | Shaka_Max | ; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT |
| 18:10.18 | Shaka_Max | ups sorry |
| 18:10.20 | Penguin | developerhealey: Goto() is the way to do that. |
| 18:11.17 | developerhealey | my_ext variable was set by read() for 3 digits... Oh, clarify, i have the dial plan set up properly to read the extension and route it properly |
| 18:11.18 | [TK]D-Fender | developerhealey: exten => s,n,Set(my_ext=${IF($["${my_ext}" = "111"]?222:${my_ext})}) |
| 18:12.06 | [TK]D-Fender | developerhealey: First mistake was having lower-case chars in your function call. |
| 18:12.10 | developerhealey | ahhh i was sooo close... |
| 18:12.18 | [TK]D-Fender | developerhealey: Next was including quotes in the result. |
| 18:12.53 | developerhealey | im a newb to asterisk, but i have experience in many other languages... Thanks for the syntax help |
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| 18:16.00 | Penguin | Or maybe yes means 'force_rport,comedia' |
| 18:16.25 | Penguin | I've got to go do some winter shopping for a while. |
| 18:19.01 | jonno11 | Penguin: Still a no go... this is pretty frustrating haha |
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| 18:20.16 | Shaka_Max | Since you are using softphones have you consider to switch to IAX2 devices? |
| 18:20.25 | Shaka_Max | or create a VPN? |
| 18:20.34 | Shaka_Max | ok no |
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| 18:22.00 | jonno11 | Sorry! IRC crashed |
| 18:24.18 | Shaka_Max | <Shaka_Max> Since you are using softphones have you consider to switch to IAX2 devices? |
| 18:24.18 | Shaka_Max | <Shaka_Max> or create a VPN? |
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| 18:28.32 | jonno11 | Shaka_Max: Hmm what do you mean? |
| 18:29.05 | Shaka_Max | instead using SIP use IAX2 |
| 18:29.28 | Shaka_Max | if you dont want to use IAX2 setup openvpn in your PBX and connect trough it |
| 18:29.28 | jonno11 | Shaka_Max: I literally have no experience with IAX2 haha |
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| 20:41.24 | jpcansa | how can I get the last call Id of a call to a certain extension? |
| 20:41.52 | jpcansa | is that stored in the AstDB by default? |
| 20:42.24 | [TK]D-Fender | First don't use the word "extension" like that, next be specific on what this "call id" is you're referring to. It could be any of about a dozen things for starters |
| 20:43.02 | [TK]D-Fender | [15:41]jpcansais that stored in the AstDB by default? <- nothing about calls automatically goes here |
| 20:43.36 | jpcansa | [TK]D-Fender: i mean the caller ID of the last call |
| 20:43.51 | [TK]D-Fender | jpcansa: CDR should have that. |
| 20:43.58 | jpcansa | and what do you mean by "the word "extension"? |
| 20:44.34 | [TK]D-Fender | jpcansa: a line in extensions.conf is an extension. |
| 20:48.17 | jpcansa | what could be an easy approach to get the Caller ID of the last call to a sip device? from the dialplan, so I can add it to a backlist in DB? |
| 20:48.50 | [TK]D-Fender | Best place is in the dialplan right before you call them |
| 20:50.06 | jpcansa | you mean add the caller ID to DB before calling the sip device? |
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| 20:52.22 | [TK]D-Fender | [15:50]jpcansayou mean add the caller ID to DB before calling the sip device? <- pretty much exactly what I said |
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| 20:59.51 | jpcansa | good thanks |
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| 21:08.37 | noko | Hello. I have avaya 9611g desktop phone and softphone with same configuration. But hardware phone can't connect to asterisk. |
| 21:09.08 | noko | Here is pcap dump of successful softphone connection http://pastebin.com/raw.php?i=Ka1p1uHD |
| 21:09.42 | noko | And here is pcap dump of avaya phone http://pastebin.com/raw.php?i=3p1NFLsB |
| 21:10.48 | noko | I don't know is avaya's "Contact:" header in sip packet is normal |
| 21:11.34 | [TK]D-Fender | use Asterisk's sip debug at CLI |
| 21:11.38 | [TK]D-Fender | "sip set debug on" |
| 21:13.46 | noko | ok, wait a second please |
| 21:14.32 | noko | [TK]D-Fender, how to grep log only for one device? |
| 21:15.33 | Penguin | Do you know how to use grep? |
| 21:15.57 | noko | :) |
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| 21:29.02 | [TK]D-Fender | checkout time, BBIAB |
| 21:38.52 | *** join/#asterisk nighty^_ (~nighty@lns-bzn-49f-62-147-170-46.adsl.proxad.net) |
| 21:43.20 | saliak | I'm trying to debug a one way audio issue still. the problem only occurs if the remote phone is called as an internal extension. that is the remote phone call call a number outside of our system just fine (two way audio), and if someone calls from the outside and dials that extension, there is 2 way audio. the remote phone is behind a NAT and i've set it up as DMZ to remove that as a variable. my firewall is on my asterisk server, so the |
| 21:43.20 | saliak | server does have non natted access to my WAN IP (phones register to that IP). firewalll has RTP and SIP open. any ideas where to look based on this behavior? |
| 21:44.28 | drmessano | Anyone know if a boot loop on a Polycom phone is something field recoverable? |
| 21:44.52 | drmessano | Is there any scenario where this isn't a call to Polycom? |
| 21:53.39 | Penguin | saliak: Asterisk has no concept of "internal extension" versus calling "from the outside," so pastebin the debug of the call with no audio. |
| 21:57.37 | saliak | Penguin: https://pastee.org/bvm59 |
| 22:03.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:04.07 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 22:04.24 | Penguin | saliak: Who is doing what in this debug? It looks like a SIP device named 8000 is trying to do something, but I'm missing some parts that show me what is going on. |
| 22:05.34 | saliak | Penguin: 8000 is trying to call 8002. 8002 is remote. 8000 is on the same local network as the asterisk server and works fine (can get calls from other internal extensions and make/receive calls without issue). |
| 22:06.46 | Penguin | I already told you that asterisk has no concept of "internal extension," so continuing to use the phrase is not helpful. |
| 22:07.05 | Penguin | Devices are devices. Extensions just execute commands. |
| 22:07.21 | saliak | Penguin: ok, sorry. |
| 22:07.44 | Penguin | The fact that SIP device 8002 is in a remote location is important. There's probably a NAT involved, and there is a definite misconfiguration. |
| 22:07.56 | saliak | Penguin: don't think that their location matters in this case? 8000 can talk to 8002 if it's on the same network |
| 22:08.07 | saliak | Penguin: I setup 8002 in DMZ for that NAT |
| 22:08.18 | Penguin | That's part of the problem. |
| 22:08.29 | saliak | Penguin: really? i thought that forwards all traffic to the phone? |
| 22:08.38 | Penguin | Not likely. |
| 22:08.50 | paulc | Anyone encountered "There was a problem writing the firmware to disk" when trying to update firmware on a Digium D70 phone? |
| 22:09.22 | Penguin | No one seems to understand DMZ, and it always seems to be a "go to" thing when there's a problem. |
| 22:09.41 | Penguin | Treat the phone just like every other PC in the network. |
| 22:09.49 | Penguin | It's just a network device. |
| 22:10.27 | Penguin | Do not configure the phone to do special things when it is behind NAT. Configure asterisk so that it knows the phone is behind NAT. |
| 22:11.04 | *** join/#asterisk daemon (staff@hashweb.org) |
| 22:11.08 | saliak | Penguin: ok. I had that before. had it setup to use STUN for nat traversal (whatever that means on a grand stream phone). at one point in all the iterations of stuff, I did have one way audio (8000->8002, but not vice versa). I think that was when I had RTP and SIP ports forwarded on it's NAT |
| 22:11.09 | daemon | hey all I want a nice girl interface for asterisk |
| 22:11.13 | daemon | asterisk-gui is deprecated |
| 22:11.15 | daemon | what are the best alternativers |
| 22:11.26 | daemon | and what is 'dhadi' I had to disable it in the compile |
| 22:11.37 | daemon | 'dahdi' |
| 22:11.46 | Penguin | I'll repeat: Do not configure the phone to do special things when it is behind NAT. |
| 22:12.01 | Penguin | Using STUN on the phone is one of those special things you are not going to be doing. |
| 22:12.16 | malcolmd | paulc: last time that happened someone had invalid firmware (bad md5)..and that also means that i've still forgotten to publish the md5 sums for the firmware on the wiki..so i'll have to add that todo. also, you might want to factory default it first and then try - since you're having some trouble. |
| 22:13.01 | paulc | @malcolmd: Yeah, I tried factory default first.. thought maybe jumping to 1_4_0 from 1_0_5 was too big a jump, so now I'm trying 1_1_0 |
| 22:13.20 | paulc | I could try downloading again too.. are the MD5 hashes on the wiki for existing builds? |
| 22:14.20 | Penguin | saliak: One-way audio is almost always a NAT misconfiguration. No-way audio seems to be a combination of NAT misconfiguration and firewall misconfiguration. |
| 22:14.33 | malcolmd | no, they're not on the wiki, that's the todo that i've just added because i forgot to do it without the todo. ;) 352c0e3a46662a3c41d30207d6c77786 - 1_4_0_0_57389_D70_firmware.eff |
| 22:15.11 | saliak | Penguin: OK, so I should remove all NAT setting on the phone, and just setup forwarding of RTP and SIP to the phone ? I thought that STUN was needed so the WAN IP was used instead of the local one for the RTP packets. |
| 22:15.23 | Penguin | saliak: No. |
| 22:16.12 | saliak | Penguin: OK, so should I setup port forwarding at all? Or you don't think that's needed either? |
| 22:16.32 | Penguin | saliak: Let me be very specific. Disable ANY AND ALL settings on the phone that are related to NAT traversal. Configure the asterisk peer entry for the phone so that asterisk knows the phone is behind NAT. No port forwarding is to be configured for an end point. |
| 22:17.12 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:17.14 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 22:17.23 | paulc | @malcolmd: Yup, same md5sum on the 1_4_0_0 file I've got downloaded.. |
| 22:17.40 | saliak | Penguin: OK. so I'll remove all forwarding. I'll remove any notion of NAT on the phone. in sip.conf on asterisk, what should I use on nat=? force_r or comedia? don't understand what those mean |
| 22:17.42 | paulc | if the phone's had a factory reset, but not setup to register anywhere - it should be ok right? |
| 22:18.02 | Penguin | saliak: Set nat=yes |
| 22:18.20 | saliak | Penguin: on 11.6 it ways it's deprecated. does that not matter? |
| 22:18.32 | La_Chota | daemon, the most popular and very maintained is FreePBX |
| 22:18.38 | malcolmd | possibly. if you've been mixing and matching configs (xml, web gui, phone menu) then that's not a good thing. |
| 22:18.57 | Penguin | saliak: Deprecated does not mean "doesn't work." |
| 22:18.59 | daemon | La_Chota, cool thank you, |
| 22:19.13 | malcolmd | paulc: how are you attempting to update the firmware? web ui? xml? |
| 22:19.16 | Penguin | saliak: It means consider updating to the current methods. |
| 22:19.20 | saliak | Penguin: ok. will try all that and check back |
| 22:19.22 | tm1000 | daemon: you dont need dahdi unless you have hardware so dont worry too much about it |
| 22:19.33 | paulc | @malcolmd: I think so far it's been all through the web.. although I was starting to dabbling with provisioning via res_digium_phones |
| 22:19.37 | daemon | tm1000, cool I am just dealing with softphones |
| 22:19.49 | paulc | @malcolmd: Trying to do the FW upgrade via the Web UI |
| 22:20.00 | tm1000 | daemon: I mean like hardware pstn, phone lines into system, not hardware phones |
| 22:24.42 | elguero | Penguin, saliak: Deprecated also means that at some point, that setting may no longer work... the suggested setting, if you want the same behavior as nat=yes, would be nat=force_rport,comedia |
| 22:24.43 | *** join/#asterisk gtjoseph (~gtj@unaffiliated/gtj) |
| 22:25.27 | elguero | Let me clarify that, may not work on future versions... so to lessen the upgrade pain, might as well start using the recommended setting |
| 22:26.15 | elguero | err, rather, recommended way of setting nat if you desire the same behavior as nat=yes |
| 22:26.26 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 22:26.34 | saliak | Penguin: it seems really confusing that a call to a number outside of our asterisk server works from the remote phone, right? in both cases (calling a number ouside of our server or 8000), do calls get relayed through the server? |
| 22:27.35 | Penguin | saliak: Also, be sure you set directmedia=no in the peer entry. |
| 22:28.33 | saliak | Penguin: ok |
| 22:28.45 | *** join/#asterisk beefcafe_ (~beefcafe@unaffiliated/beefcafe) |
| 22:28.47 | Penguin | saliak: I haven't seen all the parts of your picture yet. I don't understand your call flow, what calls where, when it is called, what happens when the call takes place... |
| 22:29.28 | Penguin | saliak: But if you configure asterisk to deal with a phone that is behind NAT, your chances of success will be increased. |
| 22:30.04 | beefcafe_ | hi, I'm using the motif channel to connect to google voice. in a normal call I would see > 0x1f13638 -- Probation passed - setting RTP source address to 10.x.x.x:16704 |
| 22:30.07 | beefcafe_ | <PROTECTED> |
| 22:30.13 | daemon | tm1000, anything other than freepbx its incompatible with our servers |
| 22:30.31 | beefcafe_ | but sometimes, there's only the second probation passed message, and the other end couldn't hear me |
| 22:30.35 | tm1000 | daemon: how is it incompatible? |
| 22:30.43 | beefcafe_ | any ideas? |
| 22:30.44 | daemon | tm1000, our servers only use postgresql |
| 22:30.47 | saliak | Penguin: |
| 22:30.57 | tm1000 | daemon: you will find no gui using postgresql |
| 22:31.03 | tm1000 | for asterisk |
| 22:31.09 | daemon | tm1000, mhmm why? asterisk its self supports postgres |
| 22:31.32 | tm1000 | daemon: because no one has coded one using postgresql and asterisk supporting postgres is irrelevant |
| 22:31.45 | tm1000 | a gui does not do everything through asterisk |
| 22:32.07 | daemon | tm1000, no but as asterisk supports postgres it is not a big jump to assume the GUI attached to it would not use a different db |
| 22:32.09 | tm1000 | otherwise it would be slow and bog down asterisk as if has to process every request everytime |
| 22:32.18 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 22:32.28 | tm1000 | daemon: Freepbx is mysql only at this time |
| 22:32.46 | tm1000 | and there are no other guis for asterisk that are free/opensource |
| 22:33.02 | *** join/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 22:33.08 | daemon | ah well time to learn cli |
| 22:33.20 | tm1000 | daemon: …..enjoy |
| 22:33.50 | tm1000 | and I only say that because you've gone from knowing nothing about dahdi or what it is, to wanting to use a gui, to now talking about CLI |
| 22:33.53 | tm1000 | time to learn *dialplan* |
| 22:33.55 | tm1000 | ~book |
| 22:33.55 | 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 |
| 22:34.01 | tm1000 | and read ^ |
| 22:34.01 | Penguin | Enjoy? Like a cold glass of Coca-Cola? |
| 22:34.28 | saliak | Penguin: yeah, definitely a better place to start! ok, so nat=yes (and I'll change it for nat=force….,comedia later if this works) and directmedia=no and clearing all the weird nat settings |
| 22:35.38 | daemon | tm1000, mhmm im not an idiot, I just need to use asterisk for a very basic softphone setup. no outside lines or even trunks. It would have been easier to use a gui and just firewall the thing off. |
| 22:35.50 | tm1000 | daemon: I never once said you are an idiot. |
| 22:36.20 | tm1000 | I showed you the book. that we have all learned from |
| 22:36.32 | Penguin | saliak: I would use nat=yes. I would probably leave it until nat=yes becomes obsoleted, but there will be no problem with changing it to the new parameter if nat=yes works. |
| 22:36.51 | beefcafe_ | anyone? |
| 22:36.57 | La_Chota | prays at the book |
| 22:37.03 | daemon | tm1000, most people in the world are not going to buy a book to setup a server for 5 softphone users ;-) |
| 22:37.19 | La_Chota | that people deserve a shot in the head |
| 22:37.31 | tm1000 | daemon: I found that book for free, |
| 22:37.34 | tm1000 | ~google |
| 22:37.34 | infobot | i guess google is http://google.com |
| 22:37.52 | Penguin | You can READ THE BOOK ONLINE. |
| 22:37.58 | La_Chota | because then that people came here to ask many many question for setup the pbx for 5 phones |
| 22:38.06 | Penguin | Or you can choose an asterisk primer such as ... |
| 22:38.09 | Penguin | ~primer |
| 22:38.09 | infobot | extra, extra, read all about it, primer is http://burner.com/asterisk-primer |
| 22:38.44 | tm1000 | Penguin: +1 |
| 22:38.51 | La_Chota | Penguin, you are Bob? |
| 22:39.18 | Penguin | Channelz made that. |
| 22:39.21 | tm1000 | "and a version is available for reading online at http://www.asteriskdocs.org/" |
| 22:39.45 | La_Chota | so you are not Bob :( I like Bobs |
| 22:40.03 | Penguin | Close, but not quite. |
| 22:40.07 | La_Chota | i mean boobs |
| 22:40.14 | La_Chota | damm again that sound terrible |
| 22:40.21 | Penguin | is a Robert, but not a Bob. |
| 22:41.26 | Penguin | tatas? |
| 22:42.32 | Penguin | tetas |
| 22:42.38 | La_Chota | Is starting to create a GUI for multitennat support based on Asterisk12 if someone want to take a look or help with the project ping me. Current State:Design and basic tests of login & tennat creation |
| 22:43.08 | La_Chota | can provide a hand diagram to explain the main goal |
| 22:43.27 | tm1000 | La_Chota: starting from scratch? |
| 22:44.18 | La_Chota | something like that, using my "distro" as GUI and reusing the code |
| 22:44.44 | X-Rob | your 'distro'? |
| 22:45.15 | La_Chota | well not a distro |
| 22:45.23 | tm1000 | La_Chota: reusing what code? |
| 22:45.34 | tm1000 | I am assuming its all code you've already written yourself |
| 22:45.50 | La_Chota | X-Rob, a GUI based on CHARISMA template on top asterisk using static realtime |
| 22:46.00 | La_Chota | yes tm1000 |
| 22:46.21 | La_Chota | so dont expect pro coding |
| 22:46.35 | Penguin | Don't expect las tetas. |
| 22:46.46 | La_Chota | JAJAJA |
| 22:47.43 | La_Chota | X-Rob, https://code.google.com/p/dm-asterisk-distro/ |
| 22:48.03 | X-Rob | So what does the web based GUI do? |
| 22:48.53 | La_Chota | tm1000, so basically the idea is have some initial web page to create tennants then to select it and then switch to configure all, endpoints, vms, confs, queues etc etc based on the selected tennant |
| 22:48.58 | La_Chota | X-Rob, configure asterisk |
| 22:49.20 | X-Rob | La_Chota: I found this on that link: Asterisk is configured to use Realtime Static, with this the SIP Peers, IAX2 Peers, Voicemail Configuration, Conference Rooms, Queues, Agents & the CDR are stored into a MYSQL Database. |
| 22:49.54 | La_Chota | yes |
| 22:50.17 | X-Rob | OK, yep |
| 22:50.24 | X-Rob | Just reading through that page |
| 22:50.35 | X-Rob | so, it pretty much configures the realtime stuff. |
| 22:50.38 | X-Rob | That's kidna handy |
| 22:50.40 | X-Rob | is it GPL? |
| 22:50.58 | La_Chota | yes |
| 22:51.17 | X-Rob | Link to the source? |
| 22:51.43 | La_Chota | i havent created a source tree yet |
| 22:52.12 | X-Rob | OK, lemme have a look inside this zip |
| 22:52.16 | La_Chota | the GUI is based on this http://http//usman.it/free-responsive-admin-template/ |
| 22:52.17 | tm1000 | La_Chota: too bad you dont support postgresql |
| 22:52.25 | La_Chota | i dont like it |
| 22:52.32 | La_Chota | and you know i love mysql |
| 22:52.34 | tm1000 | it would have made daemon's day |
| 22:52.44 | X-Rob | Postgres is -awesome- |
| 22:52.48 | X-Rob | I'm a big fan |
| 22:53.03 | X-Rob | Maybe its something I could care about. |
| 22:53.08 | X-Rob | in the BFO. |
| 22:53.11 | La_Chota | i guess is pretty easy to use postgress instead mysql |
| 22:53.43 | Penguin | You could build it to support both types and cater to more people. |
| 22:53.54 | La_Chota | well back to the topic does it sound insane tm1000 basically i want to add 1 extra colum to the asterisk db structure |
| 22:54.23 | La_Chota | adding the tennat name and then pull out only the tennant config |
| 22:54.59 | X-Rob | La_Chota: you're talking about the realtime database structure? (2 mins left on downloading this iso) |
| 22:55.01 | La_Chota | and Im trying to learn dragabble stuff in jquery to create something like a dialplan generator |
| 22:55.06 | daemon | tm1000, actually the primer you psoted is pretty clear howto do it from console im quite happy ^_^ |
| 22:55.33 | La_Chota | X-Rob, yes |
| 22:55.36 | daemon | tm1000, in honesty I only wanted a gui for time, if I can do it from scratch with less deps needed ... bonus |
| 22:55.59 | Penguin | It's so easy to do it without a GUI. |
| 22:57.06 | La_Chota | anyway im on design phase |
| 22:57.10 | La_Chota | going to eat |
| 22:58.02 | La_Chota | X-Rob, its a netinstall iso only, you can check/download the script executed in the post section |
| 22:58.08 | La_Chota | to avoid installing it |
| 22:58.19 | La_Chota | and then you can retrieve all from that script |
| 22:58.29 | La_Chota | BRB |
| 23:06.23 | X-Rob | La_Chota: http://dl.dropbox.com/u/1277237/DMDistro/DMGUIV4.tar.gz found it! |
| 23:12.10 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 23:15.17 | dar123 | does dialplan reload, updates users.conf info? |
| 23:15.29 | Penguin | ~users.conf |
| 23:15.29 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 23:16.58 | dar123 | not helpful :p |
| 23:17.12 | Penguin | I see that. It's all I've got. |
| 23:25.51 | daemon | is there anyway I can tel asterisk to do a custom SQL query to find extension details |
| 23:25.55 | SeRi | Penguin: lol |
| 23:27.53 | Penguin | dar123: If I were the one asking, I'd set my core verbosity to 10, run dialplan reload, and read the CLI to see if it read users.conf or not. |
| 23:33.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 23:34.30 | Max_E | tm1000, X-Rob this is what i done so far-->http://www.youtube.com/watch?v=4mrjvRZ1Fd4 but not sure if I choose the right path |
| 23:35.36 | Max_E | As Penguin said its easier to do multitennat via cli but you know im only want GUI or "I have a Dream" |
| 23:37.01 | X-Rob | Max_E: Looking good! |
| 23:37.19 | X-Rob | You probably need to stick it into github or something like that so you can start doing releases and stuff. |
| 23:39.11 | Max_E | with many issues i pass the github walk-trough... to much for me lol |
| 23:43.08 | *** part/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 23:44.13 | X-Rob | Max_E: if you don't use git, what do you use? svn? |
| 23:45.51 | Max_E | nothing :( |
| 23:48.10 | dar123 | got it, thanks |
| 23:52.49 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
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