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| 00:11.41 | [TK]D-Fender | Shows the comms & configs |
| 00:24.47 | rmudgett | snadge: You might be using the same auth object for incoming and outgoing authentication. That can cause unexpected failures because the realm on the incoming auth object can get a delayed default value if not supplied by the config. |
| 00:26.45 | rmudgett | You can see this by doing a "pjsip show auth" of the auth object it should be using. |
| 00:27.19 | snadge | its chan sip |
| 00:30.20 | [TK]D-Fender | Shows the comms & configs |
| 00:31.54 | snadge | yeah thats my thoughts too.. not my asterisk system, customer is selectively pasting entries from files and logs |
| 00:32.03 | snadge | ie.. not in their complete form |
| 00:32.13 | snadge | so i'll say, give us the full logs and config files |
| 00:32.18 | snadge | not just what you think we need to see |
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| 03:19.24 | Kobaz | having an odd issue. asterisk-11 |
| 03:19.25 | Kobaz | http://pastebin.com/0TDsJujD |
| 03:19.47 | Kobaz | asterisk is sending a 180 ringing after calling a peer (itsp sip trunk basically) |
| 03:20.02 | Kobaz | the 180 is coming from the itsp, and then redirected to my phone |
| 03:20.07 | Kobaz | but my phone doesn't play ringback |
| 03:20.19 | Kobaz | and it's not the phone, doesn't matter if it's a softphone, or a polycom |
| 03:20.22 | Kobaz | it doesn't play ringback |
| 03:20.50 | Kobaz | the 10.3.2.1 is nat'd ip back to my softphone |
| 03:21.34 | Kobaz | if i call a local extension, sip/399 calls sip/400 or whatever |
| 03:21.43 | Kobaz | i get the same 180 ringing, and i *DO* hear ringback |
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| 03:34.58 | Kobaz | something about progressinband |
| 03:39.04 | Kobaz | so i think i see what's going on |
| 03:43.10 | Kobaz | ;prematuremedia=no |
| 03:43.18 | Kobaz | this looks like it |
| 03:43.59 | Kobaz | Some ISDN links send empty media frames before the call is in ringing or progress state. The SIP channel will then send 183 indicating early media which will be empty - thus users get no ring signal. |
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| 04:00.34 | Kobaz | if i block 183 session progress in iptables, i get ringback now |
| 04:02.08 | Kobaz | prematuremedia=no didn't turn it off |
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| 04:13.54 | Kobaz | so the carrier is broken |
| 04:14.10 | Kobaz | it's sending 183 with no early media, so it's silent |
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| 09:02.00 | ra3burn | Trying to setup some IPv6 pjsip endpoints, and according to the Asterisk wiki I need to specify rtp_ipv6=yes for then endpoints. However, when I do this the SDP will put a IP4 (instead of IP6) in front of the IPv6 address, and the media setup will fail. |
| 09:02.11 | ra3burn | Can anyone shed some light on why this is happening? |
| 09:03.58 | ra3burn | pjsip.conf at http://pastebin.com/AGUXZbgw |
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| 09:26.12 | ruben23 | hi guys i have firewall DROP an IP address on my asterisk server for this notice --> NOTICE[3272]: chan_sip.c:23436 handle_request_invite: Failed to authenticate device 7000<sip:7000@209.126.74.10>;tag=ce4faa8c <-------------- after DROP it still appearing with this message and smae IP..it did not stop, any idea guys..? |
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| 10:56.44 | dan_j | ruben23: what firewall? |
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| 11:17.04 | wdoekes | ruben23: you probably blocked the wrong IP. the IP in that message can be forged |
| 11:17.30 | wdoekes | 'sip set debug on' and you'll see where the message really came from |
| 11:17.38 | wdoekes | s/message/request/ |
| 11:17.41 | dan_j | wdoekes: I was thinking that it's still in the connection tracking table. |
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| 11:37.38 | blinky_ | Hi all, I am looking into securing mu IAX trunks but ot really getting much help from Google. It seems that encryption is not really part of the package, is it something we need to worry about or are the voice streams safe. We have our firewall protecting the network and the server is behind that. |
| 11:39.16 | dan_j | Hi. Whats the PJSIP equivalent for SipAddHeader? |
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| 11:39.51 | dan_j | Unencypted voice streams are easily sniffable |
| 11:41.16 | blinky_ | That is what I thought but I am not finding my information on securing the streams. I have one between out provider and the box and another between our box and a remote one. Ideally I would like to secure the trunk between them both |
| 11:59.59 | dan_j | Sorry. I've never used IAX so I cant assist. |
| 12:00.46 | dan_j | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id291235.html#OutsideConnectivity_id36059999 |
| 12:00.59 | dan_j | blinky_: ^^ |
| 12:02.37 | blinky_ | Thanks for that, yeah I just found something similar that says I am able to use "encryption=aes128" as well. From what I can see the standard encryption is md5 |
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| 12:43.18 | eduzimrs | Hi guys, im stucked regard to local channels, im dialing via Manager, and im not receiving early media, is there any way to workaround this? |
| 12:44.09 | eduzimrs | it does not open rtp strem at all |
| 12:44.24 | eduzimrs | doing the same using for example x-lite, works fine |
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| 12:46.18 | eduzimrs | the other part is an Asterisk box too |
| 12:46.29 | eduzimrs | my SIP Provider |
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| 12:53.42 | eduzimrs_ | Hi guys, im stucked regard to local channels, im dialing via Manager, and im not receiving early media, is there any way to workaround this? |
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| 13:45.25 | defswork | I've got a really odd problem |
| 13:45.44 | defswork | have 2 phones, SPA525G and new Cisco 8851 |
| 13:45.59 | defswork | on the 8851 I cannot hear the inbound audio |
| 13:46.21 | defswork | they are on the same network, talking to the same server |
| 13:47.00 | defswork | problem occurs on a 8841 885 and 8861, all the old SPAs work fine |
| 13:47.59 | defswork | If I connect the 88* to different (remote) * server (very old - running 1.4) they work ok |
| 13:48.44 | defswork | I've diff'd sip debug between the 2 and cannot see any significant differences |
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| 14:39.33 | dan_j | defswork: I couldnt get the cisco phones to work properly with asterisk. i gave up in the end |
| 14:40.13 | defswork | dan_j, we have them working though against a third party * solution :o |
| 14:40.32 | dan_j | I wonder if the third party is using asterisk or something else? |
| 14:40.40 | defswork | very old asterisk |
| 14:40.41 | defswork | 1.4 |
| 14:40.51 | defswork | I have access to shell on it |
| 14:42.00 | dan_j | Maybe start off by giving 1.4 a try and see if your existing phone config works. i know it's old, but at least if you know 1.4 works, you'll have a starting point. if it doesnt work, then you'll know it's an issue with the asterisk config |
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| 14:49.22 | defswork | Our cisco sales account manager is going to get me some support |
| 14:49.47 | defswork | its most odd because an internal to internal call works |
| 14:49.59 | defswork | external to spa525 works |
| 14:50.10 | defswork | external to 88xx - no incoming audio |
| 14:50.33 | defswork | without other phones working you'd say it was a nat issue |
| 14:51.43 | [TK]D-Fender | I wouldn't say anything |
| 14:52.00 | defswork | ;) |
| 14:52.10 | [TK]D-Fender | Because at least that'd be in proportion to what we've seen |
| 14:52.58 | Samot | *Telephony features will vary by vendor platform. Consult your Cisco representative for timing availability and feature details. |
| 14:53.13 | Samot | ^ I'd ask if it has issue with old versions of Asterisk. |
| 14:53.24 | defswork | Samot, old version of * works |
| 14:53.34 | defswork | its new that I'm struggling with |
| 14:54.11 | defswork | I'll gladly share logs/debug etc.. if you'd like to see it |
| 14:54.21 | Samot | Yes. |
| 14:54.52 | [TK]D-Fender | Asterisk didn't just randomly break things |
| 14:55.18 | [TK]D-Fender | It shouldn't even be a question hat we should be looking at evidence |
| 14:55.23 | defswork | I've got a spa525 and 8851 side by side, on same network, same (as they can be) config |
| 14:55.56 | [TK]D-Fender | That is just a story and proves nothing of value for us. |
| 14:55.59 | defswork | internally only, both work aok |
| 14:56.16 | defswork | external inbound, spa525 is fine, 8851 no incoming audio |
| 14:56.31 | defswork | I've locked codecs down to alaw,ulaw only |
| 14:56.48 | [TK]D-Fender | lack of audio has nothing to do with codecs |
| 14:57.06 | dan_j | if codecs werent ok, asterisk would warn and drop the call |
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| 14:57.31 | defswork | yeah - I was just eliminating possibility that the cisco might be crapping out on GSM or something |
| 14:58.05 | defswork | and just not playing the sound - it's a linux platform, uses alsa for audio and java |
| 14:58.11 | dan_j | still have a feeling that its never going to work properly. |
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| 14:59.32 | [TK]D-Fender | I have a feeling it's going to take a long time to stop with vague descriptions and actually look at something useful |
| 14:59.49 | defswork | [TK]D-Fender, what would you like to see ? |
| 15:00.24 | [TK]D-Fender | What would you think? |
| 15:01.17 | defswork | sip debug ? |
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| 15:01.57 | [TK]D-Fender | That sounds like a kind of concrete evidence |
| 15:02.10 | [TK]D-Fender | The kind you should have opened with. |
| 15:02.33 | [TK]D-Fender | "Here's my CALL to look at, any clues as to why it doesn't have audio?" |
| 15:02.48 | [TK]D-Fender | And we might have saved 20 minutes |
| 15:05.40 | defswork | https://gist.github.com/defsdoor/f7360735325b93f95526a26d4efd1256 |
| 15:06.23 | defswork | I did a side by side walk through of the sip debug of a workign call to a spa525 and one to the 8851 though and couldn't see any signifcant difference |
| 15:06.38 | defswork | I was actually trying to avoid wasting your time ;) |
| 15:10.27 | defswork | I've connect the handsets to the server via local network and also it's external address with same outcome |
| 15:10.56 | defswork | that debug is external IP but I can do it internally also |
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| 15:21.49 | [TK]D-Fender | Peer audio RTP is at port 10.0.4.105:16420 |
| 15:21.50 | [TK]D-Fender | ^ |
| 15:21.54 | [TK]D-Fender | THERE'S your problem |
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| 15:22.00 | [TK]D-Fender | line 160 |
| 15:26.43 | defswork | [TK]D-Fender ok - but odd thing is on the SPA525 call I have exactly the same thing |
| 15:27.13 | defswork | Peer audio RTP is at port 10.0.4.42:16450 |
| 15:27.30 | [TK]D-Fender | PB the configs |
| 15:27.47 | defswork | handset configs ? |
| 15:34.16 | [TK]D-Fender | we're looking at your phone.... |
| 15:34.28 | [TK]D-Fender | so everything regarding it |
| 15:34.28 | Kobaz | [TK]D-Fender: hey |
| 15:34.44 | Kobaz | [TK]D-Fender: is there a way to tell a sip peer to not send any 183s |
| 15:36.12 | [TK]D-Fender | "notifyringing" <- start looking around there |
| 15:36.23 | [TK]D-Fender | there are otehr ones for more generic progress |
| 15:36.33 | [TK]D-Fender | I've never had to mess with this directly |
| 15:36.46 | [TK]D-Fender | Check the sample config for the specifics |
| 15:42.07 | defswork | https://gist.github.com/defsdoor/b5d37a31341e9c54e7f4ab85970e9d35 < this is the config taken from the phone - it's extension 5 that is the one I'm using for this |
| 15:50.18 | defswork | Also, I changed the extension's server ip to the internal IP - 10.0.4.25 - so the extension is connecting locally to the * server |
| 15:51.01 | defswork | theres a similar - Peer audio RTP is at port 10.0.4.105:16478 - but that IP is now the visible IP of the extension - yet still no audio |
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| 15:52.52 | [TK]D-Fender | <defswork> Also, I changed the extension's server ip to the internal IP - 10.0.4.25 - so the extension is connecting locally to the * server <- no |
| 15:52.54 | [TK]D-Fender | all wrong |
| 15:53.00 | [TK]D-Fender | the phone is on the OUTSIDE |
| 15:53.10 | [TK]D-Fender | * should reporting it will use the EXTERNAL IP it came from for this |
| 15:53.20 | defswork | well the * server is 10.0.4.25 and the phone is 10.0.4.105 |
| 15:53.27 | [TK]D-Fender | And you haven't finished my previous request\ |
| 15:53.40 | defswork | and I have localnets set to include 10.0.4 |
| 15:53.54 | [TK]D-Fender | OH |
| 15:53.58 | defswork | I pasted you the handset config ^^ |
| 15:54.01 | [TK]D-Fender | New super fail then |
| 15:54.02 | [TK]D-Fender | --- SIP read from UDP:211.37.65.115:1312 ---> |
| 15:54.02 | [TK]D-Fender | INVITE sip:07123456789@83.8.57.74 SIP/2.0 |
| 15:54.20 | defswork | no - I said I have subsequently changed it |
| 15:54.20 | [TK]D-Fender | Your phone is calling * through your EXTERNAL ADDRESS |
| 15:54.33 | [TK]D-Fender | Which woud be a hairpin NAT failure |
| 15:54.49 | [TK]D-Fender | And a Shoot-On-Sight-Offense |
| 15:55.03 | defswork | since the first sip debug, I've changed it so that the phone is talking to * over the LAN again |
| 15:55.27 | [TK]D-Fender | I think you'd better provide a new call and complete configs now.... |
| 15:55.40 | defswork | https://gist.github.com/defsdoor/cf6bcfb897df671715dfb90af7c4a6e1 |
| 15:55.56 | defswork | so that is a call from phone to * on same LAN |
| 15:56.16 | defswork | the config is identical - I just changed the DNS |
| 15:58.38 | [TK]D-Fender | I'm nott seeing any dilaplan execution or 2nd call, etc |
| 15:58.51 | [TK]D-Fender | This is half-blind |
| 15:59.11 | [TK]D-Fender | And that includes the CONFIGS I asked for |
| 15:59.51 | Kobaz | [TK]D-Fender: well i do want notifyringing |
| 15:59.56 | Kobaz | i just don't want 183 |
| 16:00.03 | Kobaz | because the carrier sends 183, and then nothing |
| 16:00.15 | Kobaz | well. 183, no media, and then a 180 |
| 16:00.26 | Kobaz | so local phones do not play any ringback |
| 16:00.38 | Kobaz | i want to nix the 183 a more proper way than just iptables -j DROP |
| 16:00.44 | Kobaz | which is working perfectly for now |
| 16:03.32 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 16:03.51 | defswork | [TK]D-Fender, https://gist.github.com/defsdoor/3756c0eb6ed13134b92bd42249a5b75a this is the dial plan execution |
| 16:04.39 | [TK]D-Fender | Can I get a SINGLE call with EVERYTHING in there so I'm not piece? |
| 16:04.42 | [TK]D-Fender | Is this so hard? |
| 16:04.57 | defswork | sure - one sec I'll combine |
| 16:05.00 | [TK]D-Fender | no |
| 16:05.13 | Kobaz | [TK]D-Fender: fun stuff right? |
| 16:05.21 | Kobaz | [TK]D-Fender: this same carrier, is now giving us one way audio |
| 16:05.25 | [TK]D-Fender | place a new call and show the ***ENTIRE*** thing. ALL of it. EVERY single piece related |
| 16:06.04 | [TK]D-Fender | this should be s single uninterrupted CLI dump, not some junk glued together |
| 16:09.06 | defswork | https://gist.github.com/defsdoor/116010715ea3c255a2733852849b16c5 |
| 16:09.29 | defswork | the dial plan stuff is in there also now - it's only a very simple dial plan |
| 16:09.39 | [TK]D-Fender | <--- Reliably Transmitting (NAT) to 10.0.4.105:5060 ---> |
| 16:09.45 | [TK]D-Fender | tthey are NOT behind NAT now |
| 16:10.22 | defswork | sip.conf:localnet=10.0.4.0/24 < I have this in sip.conf |
| 16:10.57 | [TK]D-Fender | your peer is set to think it is behind nat <------------------- |
| 16:11.11 | [TK]D-Fender | And that is NOT a complete call |
| 16:11.30 | *** join/#asterisk Y04NN (~y04nn@178.18.54.206) |
| 16:11.31 | [TK]D-Fender | And I *STILL* don't have all the configs |
| 16:11.52 | defswork | what additional configs can I get you ? |
| 16:11.57 | [TK]D-Fender | I'm getting very tired of asking for the same simple things over and over... |
| 16:12.58 | defswork | sorry, what other configs do you need - I can give you the dial plan - it's incredibly simple |
| 16:13.26 | [TK]D-Fender | <[TK]D-Fender> place a new call and show the ***ENTIRE*** thing. ALL of it. EVERY single piece related |
| 16:13.35 | [TK]D-Fender | <[TK]D-Fender> I think you'd better provide a new call and complete configs now.... |
| 16:14.17 | defswork | I've given you a complete call - with verb set to 9 |
| 16:14.23 | [TK]D-Fender | No you haven't |
| 16:14.24 | defswork | what else am I missing ? |
| 16:14.37 | [TK]D-Fender | Where's the debug for the OTHER HALF OF THE CALL? |
| 16:14.43 | [TK]D-Fender | You are NOT showing the WHOLE DAMN CALL |
| 16:15.04 | [TK]D-Fender | There seems to be problem with the word ---> COMPLETE <---- |
| 16:15.12 | [TK]D-Fender | Where is the SIP CONFIG for the phone? |
| 16:15.23 | defswork | trust me - that is everything - plus a bit after - the BYE is at line 390 |
| 16:15.35 | [TK]D-Fender | that is NOT everything |
| 16:15.41 | [TK]D-Fender | you are dialing OUT and we do NOT SEE IT |
| 16:16.01 | defswork | ok - so you want debug on the trunk also ? |
| 16:16.35 | [TK]D-Fender | TWO sides are talking. Who said you are GETTING audio from both sides? Who says you are SENDING it properly to each? |
| 16:17.02 | [TK]D-Fender | "I can't hear you" doesn't mean my hearing is defective is your mouth is SEWED SHUT |
| 16:17.21 | defswork | sorry - I thought that as it works ok to a different model handset in the same config that the the trunk side of things must be ok |
| 16:17.23 | defswork | hang on |
| 16:17.37 | [TK]D-Fender | When I say "show the ENTIRE call" I mean it |
| 16:17.39 | [TK]D-Fender | ALL OF IT |
| 16:17.48 | [TK]D-Fender | Where is the FULL configs? |
| 16:18.02 | [TK]D-Fender | Last chance on this |
| 16:18.12 | defswork | ok - before you blow a blood vessel - do you want me to cat /etc/asterisk/* into a paste ? |
| 16:18.35 | [TK]D-Fender | You didn't show SIP.CONF <---------- |
| 16:18.41 | [TK]D-Fender | TOTAL failure tto provide the basics |
| 16:18.55 | defswork | you didnt say you wanted sip.conf :| |
| 16:19.09 | [TK]D-Fender | WTF do you think sets the RULES for communication? |
| 16:20.46 | [TK]D-Fender | "my SIP phone isn't ending up with audio I think it should have" <- better look at the CONFIGS for the device |
| 16:20.59 | [TK]D-Fender | You showed the PHONE side but didn't bother to show the ASTERISK side. |
| 16:21.05 | [TK]D-Fender | As I said |
| 16:21.20 | [TK]D-Fender | [TK]D-Fender> This is half-blind |
| 16:21.20 | [TK]D-Fender | <[TK]D-Fender> And that includes the CONFIGS I asked for |
| 16:24.58 | defswork | https://gist.github.com/defsdoor/c171aba55a1cae64fdb7e2a67ebf8411 < thats sip.conf |
| 16:25.47 | *** join/#asterisk ModFather (~ModFather@unaffiliated/modfather) |
| 16:26.33 | ModFather | hi there, how can i make asterisk read/catch the presence statuses of a device ? ( lets say bria4 or X-LITE softphone, when i set it manual to busy or unavailable or on the phone ) |
| 16:28.30 | [TK]D-Fender | ModFather, Answered in #freepbx |
| 16:28.31 | [TK]D-Fender | ^ |
| 16:29.21 | ModFather | yes thank you |
| 16:36.32 | Kobaz | hah |
| 16:36.35 | Kobaz | fun stuff |
| 16:36.50 | Kobaz | TK can be a little loud and demanding, but it's worth it to stick it out |
| 16:37.24 | [TK]D-Fender | It's been 3/4 of an hour and can't get the most basic things. |
| 16:38.46 | Kobaz | it's basic for you, but. you know |
| 16:39.07 | Kobaz | when you've been asterisking for two days it's daunting to get 'everything' |
| 16:41.46 | [TK]D-Fender | It hasn't been "2 days" |
| 16:42.51 | Kobaz | well, just saying in general, dont know how much mileage he's got |
| 16:43.28 | Kobaz | so, what i think, is that level3 has an IAD in the basement |
| 16:43.48 | Kobaz | and ran a thousand feet of copper to this comm closet to deliver a smart jack |
| 16:43.58 | Kobaz | one way audio is not generally something you see on a PRI |
| 16:45.07 | WIMPy | That would be pretty hard to do. |
| 16:45.08 | [TK]D-Fender | Only time I ever had that was an HWEC bug in one of the very first firmwares for Sangoma's first A104d that had it |
| 16:45.31 | [TK]D-Fender | which was technically not a "line" or signaling issue |
| 16:45.47 | [TK]D-Fender | But is as close (if only so vaguely) to qualifying |
| 16:50.51 | igcewieling1 | Is the PRI with Earthlink? |
| 16:52.09 | igcewieling1 | IIRC, they do "VoIP PRIs". |
| 16:53.00 | [TK]D-Fender | Well he was talking about an IAD already.... |
| 16:53.18 | [TK]D-Fender | So regardless of provider a VoIP layer before going to PRI could introduce that point of failure |
| 16:55.06 | *** join/#asterisk stac (~stac@april-fools/2013/runnerup/stac) |
| 17:05.30 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 17:08.24 | *** join/#asterisk SuperNull (~supernull@96.8.2.216) |
| 17:08.46 | SuperNull | Hey guys, have you ever seen an ATA not work with a PBX for Caller Id but it works fine on a handset ? |
| 17:09.07 | *** join/#asterisk coldfront (~coldfront@162.243.40.229) |
| 17:09.21 | [TK]D-Fender | what do you mean "handset"? |
| 17:09.25 | [TK]D-Fender | What ATA? |
| 17:09.55 | [TK]D-Fender | How does an ATA work with a handset, but NOT a PBX? When is an ATTA not being used with a PBX? |
| 17:10.24 | SuperNull | so handset being standard unpowerd POTS phone .. |
| 17:10.33 | SuperNull | the ATA is actually in a GPON ont in this case.. |
| 17:11.25 | SuperNull | i have seen this before not sure if the devices are just not following the standards or .. what but i have had to use overly expensive TA900 series to resolve this.. just not sure why this would be |
| 17:12.09 | *** join/#asterisk Y04NN (~y04nn@178.18.54.206) |
| 17:13.41 | *** join/#asterisk defsdoor (~andy@cpc35-sutt4-2-0-cust184.19-1.cable.virginm.net) |
| 17:15.04 | [TK]D-Fender | we'd have to look at the actual device & calls |
| 17:15.58 | SuperNull | someone is suggesting FSK timing is the issue hm, not sure if i can even change that honestly. |
| 17:16.10 | SuperNull | yeahhh probably with a scope or something to see when we have ringing and FSK burst |
| 17:28.22 | SuperNull | heh. i can only change if its telcordia or ETSI based.. aka no options.. lel. |
| 17:43.50 | *** join/#asterisk libardi (~libardi@187.64.229.96) |
| 17:48.55 | Kobaz | igcewieling1: level3 |
| 17:49.13 | Kobaz | they have fiber to a building |
| 17:49.35 | Kobaz | i'm just really not sure why they insisted on pri, i doubt they have any copper back to the CO |
| 17:49.49 | Kobaz | but PRI was all that was available |
| 17:50.00 | Kobaz | so for $350 a month plus minutes, you get a PRI |
| 17:50.16 | igcewieling1 | that's not a bad price |
| 17:50.19 | Kobaz | or for $575 a month, you can get a data circuit, and then roll voip on it |
| 17:50.28 | Kobaz | but there was no in between |
| 17:51.09 | igcewieling1 | Kobaz: in my experience (which is from a few years ago), SIP handoff required significant minute commitments. |
| 17:54.26 | *** join/#asterisk qakhan (~qakhan@50-204-254-11-static.hfc.comcastbusiness.net) |
| 17:54.50 | Kobaz | so weird |
| 18:06.26 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 18:09.42 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 18:21.53 | *** join/#asterisk tristero (~al.f.zero@unaffiliated/transfinite) |
| 18:40.17 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 18:44.03 | *** join/#asterisk DoYouNo (~DoYouKnow@unaffiliated/doyouknow) |
| 18:51.17 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-yuffrzcaejlmqrfw) |
| 18:51.17 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 18:55.26 | Kobaz | oh man |
| 18:55.27 | Kobaz | so |
| 18:55.34 | Kobaz | [TK]D-Fender: the one way audio is coming from me :( |
| 18:55.52 | Kobaz | [TK]D-Fender: on our adtran ta904, i tried to fix the spurrious 183 issue, and did a ringback override on it |
| 18:56.01 | Kobaz | and for whatever reason, that creates one way audio |
| 19:04.36 | *** join/#asterisk karelk (~karel@31.10.149.26) |
| 19:08.03 | *** join/#asterisk djadk (~djadk@148.0.46.251) |
| 19:08.20 | *** join/#asterisk Janos (~Janos@190.171.120.15) |
| 19:10.12 | Janos | hey there, got a question, I have a context that includes other contexts, is there any way to modify the caller id of everything that enter that context before it's sent to the correct extension on one of the included contexts ? |
| 19:11.00 | Janos | maybe another question would be, is there a way to excute code for everything that enters a context before it's sent to the matching extension ? |
| 19:12.40 | Kobaz | yes and no |
| 19:12.49 | Kobaz | you can do it. but not with includes |
| 19:12.57 | Kobaz | set up a _X. in your 'main' context |
| 19:13.06 | Kobaz | do all your generic handling there |
| 19:13.22 | Kobaz | and then do a goto, to another context, that has the includes |
| 19:13.35 | Kobaz | so basically the same layout, but you can't use includes the whole way |
| 19:13.43 | Kobaz | you need an upper level context to jump into first |
| 19:13.53 | Janos | Kobaz, thanks for the advice, sounds like a good alternative, will do |
| 19:14.06 | Kobaz | so what i usually do is like context generichandler { dostuff() goto _generichandler, s, 1; } |
| 19:14.19 | Kobaz | just rename your current context to be prefixed with an _ |
| 19:14.28 | Kobaz | and then the new one is the same name as the old one |
| 19:14.35 | Kobaz | so your other code that uses it... nothing has to change |
| 19:15.08 | Kobaz | and if you're not using ael, highly recommended |
| 19:15.15 | Kobaz | it'll majorly clean up your dialplan |
| 19:16.52 | Janos | Kobaz, great tips, thanks, I normally use lua as dialplan and I love it, but in this specific case, the dialplan is already build with the traditional extensions.conf |
| 19:23.05 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:8126:1bc8:4ff4:c795) |
| 19:28.35 | *** join/#asterisk siegmarb (5b2fd743@gateway/web/freenode/ip.91.47.215.67) |
| 19:29.37 | siegmarb | Some sip invites are ignored by asterisk. Who to blame and how to fix? I suspect something with MTU /package size: this call is accepted: http://paste.debian.net/908482/ this call is not: http://paste.debian.net/908483/ |
| 19:34.24 | *** join/#asterisk DoYouNo (~DoYouKnow@unaffiliated/doyouknow) |
| 19:39.20 | Samot | sipbalance4-2.1und1.de.5060 > askoziapbx.local.5060: SIP, length: 1368 |
| 19:39.41 | Samot | <PROTECTED> |
| 19:40.00 | Samot | The Record Routes clearly show these calls coming from two separate sources. |
| 19:40.26 | Samot | Do you have a peer in Asterisk for both hosts? |
| 19:41.10 | siegmarb | howto check? |
| 19:42.02 | Samot | Look at your SIP or PJSIP settings. |
| 19:42.19 | Samot | Whatever channel driver you are using to connect to provider. |
| 19:43.31 | siegmarb | shouldt the output of sip show peers be fine? |
| 19:44.11 | Samot | Are you using Chan_SIP? |
| 19:44.16 | siegmarb | yes |
| 19:44.22 | Samot | Then each peer support one host. |
| 19:44.38 | Samot | If they are sourcing calls from multiple IPs, you need a peer for each of them. |
| 19:44.57 | siegmarb | ok, i do not have a peer for any of the above source hosts |
| 19:45.16 | siegmarb | however asterisk is accepting call from peer sipbalance4-2.1und1.de |
| 19:45.25 | *** join/#asterisk klow (~klow@66.114.139.162) |
| 19:47.07 | [TK]D-Fender | Where do we see * SIP debug for these calls? |
| 19:50.59 | siegmarb | sip show debug is not even showing this call |
| 19:51.02 | siegmarb | because it ignores it |
| 19:51.10 | siegmarb | sorry sip set debug on |
| 19:55.19 | Samot | But it shows the "good" one |
| 19:55.26 | Samot | So show us the good one. |
| 19:55.58 | siegmarb | whats the purpose if you cant compare? |
| 19:58.06 | siegmarb | http://paste.debian.net/908493/ |
| 19:58.09 | siegmarb | thats a good oone |
| 19:59.02 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:a06f:7a85:6008:ef33) |
| 19:59.54 | *** join/#asterisk ppd1990 (~mfederle@p5B122B17.dip0.t-ipconnect.de) |
| 20:05.15 | Samot | No matching peer for '+491791194767' from '212.227.67.131:5060' |
| 20:05.22 | Samot | Show your sip.conf, mask any secrets. |
| 20:05.31 | Samot | ~pb |
| 20:05.31 | infobot | [pastebin] a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:05.49 | siegmarb | this is the call that is going through, so thats working |
| 20:08.02 | Samot | Let's see how you have your peers configured. |
| 20:08.40 | ppd1990 | hello. I have a question regarding remote bridging. If Asterisk chooses to remotely bridge two channels, does this mean I will have a problem because my phone is behind a NAT (Ports forwarded to Asterisk Server)? How can I disable this with PJSIP? In my current Asterisk 14 setup I can only get two-way audio if I call to outside. Incoming calls will only transmit audio from outside to inside the LAN. Any tips would be greatly appreciated |
| 20:09.23 | siegmarb | Samot: http://paste.debian.net/908498/ |
| 20:12.34 | Samot | host=sipbalance4-4.1und1.de |
| 20:13.01 | Samot | Here's what you need to do.. |
| 20:13.14 | Samot | Contact or go to 1und1.de's website |
| 20:13.19 | Samot | Or their tech support.. |
| 20:13.35 | Samot | Find out what IP/hosts you need to allow for them to send calls from. |
| 20:14.40 | siegmarb | why is this call accepted without beeing listed as peer? sipbalance4-2.1und1.de ? |
| 20:15.42 | siegmarb | shit! because it's part of the first one sip.1und1.de |
| 20:15.43 | WIMPy | The common ones are sipbalance{1-10}-{1-4}.1und1.de. |
| 20:16.08 | siegmarb | isnt there an option to allow all? |
| 20:16.29 | Samot | sipbalance4-4.1und1.de and sipbalance4-2.1und1.de |
| 20:16.34 | Samot | Resolve to the same IP |
| 20:16.34 | WIMPy | allowguest |
| 20:16.38 | igcewieling1 | siegmarb: there is only an option to 'allow all including hackers' |
| 20:16.53 | Samot | sipbalance5-3.1und1.de does not |
| 20:16.55 | Samot | That's is why |
| 20:17.28 | Samot | Asterisk resolves the host names.. |
| 20:17.36 | Samot | Two host names that resolve to the same IP...yeah.. |
| 20:17.41 | WIMPy | There are no duplicate IP in those 40 hosts. |
| 20:18.23 | Samot | 212.227.67.204 is from Germany (DE) in region Western Europe |
| 20:18.23 | Samot | Input: sipbalance4-2.1und1.de |
| 20:18.36 | Samot | 212.227.18.204 is from Germany (DE) in region Western Europe |
| 20:18.36 | Samot | Input: sipbalance4-4.1und1.de |
| 20:18.42 | Samot | Oh crap. |
| 20:18.51 | WIMPy | Not the same IP at all. |
| 20:18.53 | Samot | Stupid octect. |
| 20:19.06 | Samot | Yes. |
| 20:19.14 | Samot | Third orctect is different. |
| 20:19.18 | Samot | whatever. |
| 20:19.24 | Samot | Add the host. |
| 20:19.26 | siegmarb | so this sounds like stupid from my provider |
| 20:19.36 | siegmarb | to nost add all ips to sip.1und1.de |
| 20:19.51 | WIMPy | If you use chan_sip you have to add all the hosts. |
| 20:19.53 | siegmarb | any time they change, i have to keep up |
| 20:20.25 | Samot | 2:44:44 PMÂ <Samot>Â If they are sourcing calls from multiple IPs, you need a peer for each of them. |
| 20:20.44 | siegmarb | yes they do, looks like round robin |
| 20:20.54 | *** join/#asterisk jozza (~chatzilla@unaffiliated/jozza) |
| 20:20.55 | siegmarb | or probably depends on the area the call is originating |
| 20:21.17 | WIMPy | Don't they send calls from multiple hosts any more? |
| 20:21.33 | siegmarb | any other hack or workaround available? |
| 20:21.43 | WIMPy | no |
| 20:22.00 | WIMPy | Just create 40 peers and you should be safe. |
| 20:22.15 | WIMPy | In fact the last 2 don't actually exist. |
| 20:23.41 | WIMPy | Looks like they did away with extreme high availability. |
| 20:28.40 | igcewieling1 | Unless you want to handle authentication yourself as part of the dialplan ( a VERY bad idea), then do what Wimpy said. |
| 20:32.11 | Samot | Man,I would pass on a provider with 40 peers. |
| 20:35.07 | WIMPy | http://voice.yeti.dk/sip.1und1 |
| 20:36.15 | siegmarb | no i added all 40 peers |
| 20:36.24 | siegmarb | still the invites are ignored |
| 20:38.09 | igcewieling1 | siegmarb: I'm pretty sure they should be rejected, not ignored. |
| 20:39.45 | siegmarb | do i have to add the hostnames or the ips? |
| 20:40.00 | siegmarb | because i added the hostnames but see that askozia/asterisk rejected |
| 20:40.02 | siegmarb | indeed |
| 20:40.37 | WIMPy | Hostnames are fine. See the URL I posted. |
| 20:44.23 | siegmarb | i did. |
| 20:44.30 | siegmarb | no change. I still think its the package size. |
| 20:45.13 | Samot | Show the call |
| 20:46.03 | Samot | asterisk -rvvvvvvvvvv |
| 20:46.08 | Samot | sip set debug on |
| 20:46.15 | Samot | open another SSH session. |
| 20:46.23 | Samot | Run the tcpdump command.. |
| 20:46.25 | Samot | Make a call |
| 20:46.40 | Samot | Post both the output from Asterisk and the tcpdump. |
| 20:46.59 | Samot | I don't care if the only output you show from asterisk is those two commands and a console prompt. |
| 20:48.22 | siegmarb | http://paste.debian.net/908507/ |
| 20:48.51 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:a06f:7a85:6008:ef33) |
| 20:50.17 | WIMPy | Ah. There we have the many hosts again. |
| 20:51.46 | siegmarb | http://paste.debian.net/908509/ |
| 20:52.24 | siegmarb | not a single INVITE in the sip log |
| 20:52.49 | siegmarb | it's something with the fragmented package or the size of the package i guess |
| 20:53.02 | WIMPy | Do you have any IP filters active? |
| 20:53.05 | Samot | Retransmitting #1 (NAT) to 212.227.18.200:5060: |
| 20:53.09 | Samot | Not behind NAT. |
| 20:53.15 | Samot | Or I would think they are not. |
| 20:53.38 | siegmarb | just regular MASQUERADING between lan/wan |
| 20:54.35 | WIMPy | Was the tcpdump from the host running Asterisk? |
| 20:54.54 | siegmarb | yes |
| 20:54.54 | igcewieling1 | does wireshark / tcpdump show the packets actually arriving? |
| 20:55.00 | igcewieling1 | .me looks at WIMPy |
| 20:55.16 | WIMPy | Any IP filters in sip.conf? |
| 20:55.35 | WIMPy | donates a / to igcewieling1 :-) |
| 20:55.59 | siegmarb | i only have this in place |
| 20:56.01 | Samot | Retransmitting #4 (NAT) to 212.227.18.200:5060: |
| 20:56.03 | siegmarb | localnet=192.168.0.0/24 ; manual attributes tos_sip=cs3 tos_audio=ef localnet=10.8.0.0/24 |
| 20:56.34 | WIMPy | Yes, Those retransmissions look related. |
| 20:57.17 | WIMPy | And you don't have anything else listening on the SIP port, either? |
| 20:57.19 | Samot | What is the local IP of the PBX? |
| 20:57.29 | siegmarb | 192.168.0.101 |
| 20:57.56 | siegmarb | no, it's only enable sip alg so that i do not have to forward any ports from outside to the pbx |
| 20:57.56 | Samot | 21:47:29.347615 IP sipbalance2-3.1und1.de.5060 > askoziapbx.local.5060: SIP: INVITE sip:4986219883088@192.168.0.101:5060 SIP/2.0 |
| 20:58.16 | Samot | SIP: INVITE sip:4986219883088@192.168.0.101:5060 SIP/2.0 |
| 20:58.20 | WIMPy | Ohoh |
| 20:58.30 | igcewieling1 | sip alg? |
| 20:58.32 | Samot | That's from the tcpdump. |
| 20:58.54 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:94aa:516e:9b48:b8dc) |
| 20:59.17 | WIMPy | SIP ALG is usually causing much more harm than good. |
| 20:59.49 | siegmarb | but i do not want to forward ports from outside |
| 20:59.52 | igcewieling1 | I've never heard of SIP ALG helping anything with regards to Asterisk.' |
| 20:59.59 | igcewieling1 | you don't need to forward ports from the outside. |
| 21:00.15 | WIMPy | I don't know what kind of firewall you hav, but that's not neccessary, usually. |
| 21:00.26 | siegmarb | its linux with iptables |
| 21:00.48 | siegmarb | and i had to enable nf_nat_sip |
| 21:00.51 | igcewieling1 | If asterisk generates an outbound packet the reply packets will be natted correctly. You might miss the first second of audio as the nat translations for audio need to be setup |
| 21:01.19 | WIMPy | If you want to register pones from the internet, you need to forward the SIP port. Otherwise it should work without any forwarding. |
| 21:01.44 | igcewieling1 | much the same as browsing the web, but since it is udp the timeouts are often 30 seconds instead of hours like with TCP. |
| 21:01.50 | WIMPy | rmmod nf_nat_sip |
| 21:02.17 | igcewieling1 | wow. how many hours have you wasted because sip alg was enabled? |
| 21:02.19 | siegmarb | ok, will give it another try |
| 21:02.21 | WIMPy | Yes. You might want to look at the timeouts or enable keepalive in Asterisk. |
| 21:03.10 | siegmarb | JESUS! its ringing |
| 21:03.16 | igcewieling1 | a symptom of nat translations timing out is that outbound calls work. inbound calls only work for 1 min after asterisk registers with the provider. |
| 21:03.44 | siegmarb | thank you so much guys! :p |
| 21:04.19 | siegmarb | but i have to test this, in the past i had issues, beeing available from outside after a while without sip alg |
| 21:10.10 | Samot | SIP ALG is satan spawn. |
| 21:10.32 | WIMPy | d/ALG / |
| 21:10.42 | Samot | I have never found SIP ALG make anything better. |
| 21:11.54 | igcewieling1 | siegmarb: I suspect your previous problem was with nat timeouts. |
| 21:17.33 | Samot | WIMPy would prefer we all go back to PRI. |
| 21:17.43 | Samot | TDM is Jesus's gift. |
| 21:18.05 | WIMPy | likes things that work. |
| 21:18.27 | WIMPy | likes things that simply work even more. |
| 21:20.08 | igcewieling1 | hugs our PRI |
| 21:20.19 | Samot | I've never had a real issue with SIP. |
| 21:21.08 | Samot | I've dealt with NAT issues and that crap.. |
| 21:21.25 | Samot | But 98% of the time it was Residential and a shitty router. |
| 21:21.35 | Samot | With things like SIP ALG and SPI firewall, etc. |
| 21:31.15 | siegmarb | indeed, without sip alg i have no audio after 12 minutes |
| 21:31.21 | siegmarb | call is ongoing but i do not hear anything anymore |
| 21:35.46 | igcewieling1 | siegmarb: disable session timers and see if it helps |
| 21:36.21 | igcewieling1 | also, you should be able to set a UDP NAT translation timeout for port 5060 in iptables or maybe sysctl. |
| 21:37.03 | igcewieling1 | you have directmedia=no, correct/ |
| 21:38.58 | siegmarb | correct |
| 21:39.55 | siegmarb | disabling session timers did the trick |
| 21:40.09 | siegmarb | that options sounded like a nice to have |
| 21:40.22 | igcewieling1 | try rtpkeepalive |
| 21:40.45 | siegmarb | i give it now another round of 15 minutes with disabled sessiontimers |
| 21:40.46 | igcewieling1 | siegmarb: session timers sound great, but they have never worked for me. |
| 21:41.04 | siegmarb | step by step to find the wrong settings |
| 21:41.35 | WIMPy | has session-timers on everywhere. |
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| 21:48.25 | igcewieling1 | WIMPy: I assume it is something with our setup or hardware/software mix. |
| 21:49.15 | ppd1990 | guys, is there a way to set "external_media_address" to a domain (e.g. dyndns) in pjsip.conf? |
| 21:51.24 | Samot | No. |
| 21:51.29 | Samot | Not at the moment. |
| 21:51.41 | Samot | external_media_address only supports IPs. |
| 21:52.23 | Samot | I think file said that was a feature add happening.. |
| 21:52.34 | Samot | Not a 100% though. |
| 21:59.45 | ppd1990 | is it even strictly necessary to use this for a non-static IP NAT situation? I have set up such a configuration and I can receive and make calls |
| 22:04.09 | ppd1990 | or is this because my ITSP just automatically reuses the IP from the registration?! |
| 22:06.39 | igcewieling1 | the reason registration exists is to tell the remote server what your IP and port are. |
| 22:08.12 | ppd1990 | ah. So in this case there is no need for SDP rewrites? |
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| 22:21.38 | igcewieling1 | ppd1990: Asterisk fixes up the SDP before sending it. |
| 22:21.56 | igcewieling1 | assuming you have things like nat=yes and externip= |
| 22:26.12 | ppd1990 | igcewieling1: and if I don't have those? Is it then supposed to work over NAT? (assuming a dynamic host) |
| 22:26.39 | ppd1990 | and also assuming that direct_media is off |
| 22:27.08 | igcewieling1 | dynamic ips make it more complicated if they change often. |
| 22:27.52 | igcewieling1 | if you don't set externip / externhost then asterisk won't know what ip to use in the SDP. |
| 22:30.38 | ppd1990 | I'm using the phone right now on a setup that does involve dynamic ips and everything seems to work. Is this a mere coincidence or does the ITSP just use the SDP sender's ip if it can't route to the local ip of the Asterisk server? |
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