| 00:00.26 | bkuhn | wyoung: no, I have all of them set to the same codec (both in the case where I get the strange breakup and when I don't) |
| 00:00.52 | bkuhn | (and I confirmed all the channels report the same codec) |
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| 00:11.52 | Ellenor | hugs wyoung and bkuhn |
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| 06:59.58 | _boot | hi all, does anyone know why Asterisk might refuse SRTP calls with a 488 Not acceptable here? I've got the phone set up with mandatory SRTP, and encryption=yes in sip.conf, and Asterisk can successfully call the phone... It's just the phone can't make outbound calls |
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| 07:30.55 | _boot | I've got the debug logs of an inbound call (successful) http://pastebin.com/MSucLzmq and an outbound call (failed, 488 Not Acceptable Here) at http://pastebin.com/wtB9xfdE - 2001 is the extension which i'm trying to enable srtp for |
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| 08:12.09 | stefan27 | before my channel SIP_B_0 executes Dial(SIP/A) I want to manipulate the From header (sip:44@boo.com) in the outgoing invite to A. However I don't want to do Set(CALLERID(num)=44) on channel B because I don't want consumers of AMI channel events to know about the callerid 44... is it bad to tamper the from header directly without using CALLERID? |
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| 09:58.53 | Samot | _boot: FIx your baresip client codec settings |
| 09:59.36 | _boot | fix how, what is wrong with them? |
| 09:59.43 | _boot | (it works on calls from Asterisk -> baresip) |
| 09:59.51 | Samot | Your presenting a ton of codecs you don't need. |
| 10:00.12 | Samot | You're not using G722 or g726 I'm pretty sure. |
| 10:01.20 | Samot | Limit baresip to use g711 and see what happens. |
| 10:04.49 | _boot | http://pastebin.com/8wE0GhNy |
| 10:05.07 | _boot | still hitting a Not Acceptable Here |
| 10:07.28 | Samot | _boot: set the verbose to 10 so show how the call is being handled. |
| 10:07.40 | _boot | core set debug 10 right? |
| 10:08.29 | _boot | oh wait no theres a separate one for verbose :) |
| 10:10.17 | Samot | core set verbose 10 |
| 10:11.29 | _boot | http://pastebin.com/fY3DGt9G - i set both debug and verbose to 10 but the only difference i notice is 'Using SIP RTP CoS mark 5' |
| 10:12.21 | Samot | That's not even a complete call with sip debug let along verbosity |
| 10:12.27 | Samot | That's not even a complete call with sip debug let alone verbosity |
| 10:12.40 | _boot | but that's all i'm getting |
| 10:30.05 | Samot | Do you have this setup right for TLS? |
| 10:30.35 | _boot | i believe so, tls signalling seems to be working and asterisk is able to call baresip with tls + srtp |
| 10:31.18 | _boot | if i turn off srtp completely tls still works and calls can be made from baresip |
| 10:32.05 | Samot | Is the TLS cert public or private? Ie did you self gen? |
| 10:32.52 | _boot | its not self-signed |
| 10:33.50 | Samot | What was the common name used on the cert? |
| 10:33.58 | _boot | sip.awful.name |
| 10:35.53 | Samot | Is that the real FQDN? |
| 10:36.12 | _boot | i believe everything is in order, openssl s_connect doesn't throw any nasty surprises |
| 10:36.15 | _boot | yep |
| 10:37.14 | _boot | s_client *whoops |
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| 10:39.43 | Samot | Then show an unsanitized call. |
| 10:39.54 | _boot | sure thing |
| 10:41.50 | _boot | http://pastebin.com/6qhjjGxU |
| 10:56.48 | Samot | So have you tried this on something else beside baresip? |
| 10:57.04 | Samot | Could be something baresip is sending that Asterisk doesn't like when it's using TLS. |
| 10:57.43 | _boot | i don't have anything else to try it on here, i'll have to find something |
| 11:01.33 | _boot | okay, i'll come back later and try setting it up on a hardware phone, thanks for your time :) |
| 11:03.16 | Samot | _boot: So you're securing internal calls for a reason right? |
| 11:04.37 | _boot | srtp and tls is for the stuff outside the LAN |
| 11:04.56 | Samot | So just for endpoints that are remote. |
| 11:05.17 | _boot | yep, just testing between 1001 and 2001 at the moment |
| 11:07.30 | Samot | OK. Just making sure. Some will think that TLS is a 100% secure thing for calls. |
| 11:07.49 | Samot | They don't get it's endpoint <--> PBX |
| 11:08.20 | _boot | yeah |
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| 12:00.10 | wyoung | endpoint <--> sneaky server <--> PBX |
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| 13:56.46 | Ahmadi | Hi |
| 13:57.56 | Ahmadi | I have multiple lines(all are incoming calls), after conference them, they can't talk together . seems that conference is muted. |
| 13:58.01 | Ahmadi | Anyone can help me? |
| 13:58.34 | Ahmadi | Is there any option or setting that cause the problem? |
| 13:59.12 | Ahmadi | Transfer,Call-In, Call-Out of PBX work correctly, just conference is muted |
| 14:01.22 | wyoung | could be a firewall issue |
| 14:01.31 | wyoung | do you have directmedia enabled? |
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| 14:01.48 | wyoung | reinvite? |
| 14:01.54 | Ahmadi | wyoung, what is directmedia? |
| 14:03.22 | Ahmadi | When i transfer CallA to a number, voice work. but in conference all things is silent. If its firewall. must block all voices not only conference |
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| 14:08.24 | Ahmadi | wyoung, Got it, I'm checking "reinvite" |
| 14:26.38 | wyoung | Ahmadi: it may block when adding to the conference as the media path may change |
| 14:26.47 | wyoung | unless you specific that is doesn't in your config :) |
| 14:27.47 | Ahmadi | after adding "canreinvite = yes" to my "sip.conf" and rebooting , also audio stream dont work |
| 14:28.11 | wyoung | change it to no |
| 14:28.30 | wyoung | are you behind NAT / firewall? |
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| 14:28.56 | wyoung | if so you may need to set the localnet and externip as well |
| 14:29.05 | Ahmadi | i disabled firewall . don't solve the problem |
| 14:29.24 | wyoung | are you using NAT? or does your asterisk box have a valid IP? |
| 14:29.30 | wyoung | valid external IP even |
| 14:30.00 | Ahmadi | wyoung, no, i don't using NAT. asterisk and client(extension) both are in local network |
| 14:30.05 | Ahmadi | without any firewall |
| 14:31.05 | wyoung | Ahmadi: and all parties in the conference are in your local network too? |
| 14:31.43 | wyoung | is it all SIP? or are you using DAHDI, IAX or something else? |
| 14:32.58 | bkuhn | wyoung: any more thoughts about that problem I was asking about last night? |
| 14:32.59 | Ahmadi | wyoung, parties come from mobile network, i make call to parties with my sip client(extension). when both answer the call. make conference them. after conference them, they can't talk together |
| 14:33.54 | Ahmadi | wyoung, My client is a SIP Client. just SIP protocol |
| 14:33.54 | wyoung | bkuhn: My memory doesn't extend past 12 hrs ago |
| 14:34.02 | bkuhn | :) |
| 14:34.06 | Ahmadi | ;) |
| 14:34.39 | wyoung | Ahmadi: ok but you are going outside of your local network to contact your ISTP |
| 14:34.57 | bkuhn | So, my problem is I have two SIP calls, this particular SIP client has a lot of breakup, it's very regular sounding, like timing. However, on calls where it says : " "Locally bridging <bad_Channel> and <channel>"" the bad_channel no longer has breakup. |
| 14:35.08 | bkuhn | I was wondering if there's a way to force Asterisk to do that "Locally bridging" |
| 14:35.15 | wyoung | bkuhn: which was lunch time, I had chicken pho noodle soup, it was nom. |
| 14:35.41 | wyoung | bkuhn: hmmm, no idea |
| 14:35.50 | bkuhn | you asked if I was trascoding, but I'm not |
| 14:35.57 | wyoung | ah ok |
| 14:36.02 | wyoung | yes I did ask that |
| 14:36.37 | wyoung | Ahmadi: have you set localnet and externip? |
| 14:36.39 | Ahmadi | wyoung, Sure. the GSM Mobile Network is out of my network. I call with my out-going trink |
| 14:36.57 | Ahmadi | *trink=trunk |
| 14:37.17 | wyoung | Ahmadi: and your outgoing trunk is not in your local network, so when you setup a conference it is possible that the media is being sent to a stupid IP address instead of your actual external IP address |
| 14:37.41 | wyoung | Ahmadi: what devices are you using to initialise the conference call/. |
| 14:37.54 | Ahmadi | wyoung, Then why transfer call work correctly? im confused |
| 14:38.27 | Ahmadi | wyoung, When i transfer a call, all media work correctly , both parties call talk together |
| 14:38.30 | wyoung | Ahmadi: *shrugs* |
| 14:38.46 | wyoung | anything logged in asterisk -rvvvvvvvvdddddddd |
| 14:38.48 | wyoung | ? |
| 14:38.54 | Ahmadi | What is different between transfer and conference |
| 14:39.43 | wyoung | depends on the phone you are using |
| 14:40.23 | Ahmadi | wyoung, Can't debug it on asterisk log? |
| 14:40.49 | wyoung | a phone will usually do a transfer by asking the asterisk server to do it. some phones may try to call the party directly and join the call directly when using 3 way conferencing |
| 14:40.53 | Ahmadi | Can i debug it on asterisk log? |
| 14:41.22 | wyoung | Ahmadi: it would be easier to do it using the console, but yeah you can do it on the log you just need to increase verbose and debug levels |
| 14:42.00 | wyoung | Ahmadi: if you set localnet and externip then the asterisk server will rewrite the SIP packets leaving the localnetwork with the correct external IP |
| 14:42.34 | Ahmadi | Im not familiar with localnet/externip |
| 14:42.54 | wyoung | google |
| 14:43.08 | Ahmadi | OK. Thanks |
| 14:43.14 | Ahmadi | Googling... |
| 14:43.19 | wyoung | it tells asterisk when to do rewriting of sip packets |
| 14:43.29 | wyoung | of course address that is |
| 14:43.31 | wyoung | source* |
| 14:43.43 | Ahmadi | Is audio media also a sip packet? |
| 14:43.53 | wyoung | RTP |
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| 14:44.47 | wyoung | SIP sets up where to send the RTP, if you are behind NAT and you don't set externip and localnet then SIP can tell RTP to connect to your local IP address isntead of your Internet IP address |
| 14:45.23 | wyoung | which wont work unless you have setup a VPN between the two sites ;) |
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| 15:17.22 | xpheres | hello, will be ever see video negotiation capability in asterisk? |
| 15:19.18 | [TK]D-Fender | huh? |
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| 15:30.48 | ruied | Hi. I'm configuring queues.conf and I have "number sip/22,1" and "number sip/23,2" . When sip/22 is on a call it receives also the second call that I would like to go to sip 23. My strategy = ringall , but it goes always to sip/22 |
| 15:32.01 | ruied | sip sho queue shows sip/22 (Ring+InUse) when the second call starts, but I would like that it goes to sip/23 with a penalty 2 |
| 15:32.04 | [TK]D-Fender | if it's ringall it'll always go to both |
| 15:32.14 | [TK]D-Fender | unless you tell it NOT to ring the busy one |
| 15:32.48 | ruied | it just rings sip/22 |
| 15:33.47 | [TK]D-Fender | well you've assigned a penalty to them.... |
| 15:33.51 | [TK]D-Fender | so they are not the same... |
| 15:34.08 | [TK]D-Fender | Why would you think that the the 2nd one would get chosen? |
| 15:36.25 | ruied | hmm, because the first one was In use, I thought than it jumped to he second penalty... thats what I was thinking.... |
| 15:37.11 | [TK]D-Fender | There is another file that controls changing penalty levels.... |
| 15:37.20 | file | controls everything |
| 15:38.03 | [TK]D-Fender | "One file to control them all, and in the compiler bind them." |
| 15:40.48 | ruied | queuerules right? |
| 15:42.05 | [TK]D-Fender | The name was pretty clear, wasn't it? |
| 15:42.19 | [TK]D-Fender | ... and the description from the sample config.... |
| 15:42.34 | ruied | yep :) |
| 15:47.03 | [TK]D-Fender | You could also look at telling * to NOT call that first one when busy.... |
| 15:47.24 | [TK]D-Fender | The sample configs give you some pretty big hints on that |
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| 15:48.48 | ruied | yes, but I would like that the first one to notice that there is another incoming call... |
| 15:49.27 | ruied | hmm.... |
| 15:49.48 | [TK]D-Fender | So what is your actuall full targeted scenario? |
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| 15:53.42 | wyoung | woooo!!! [TK]D-Fender!!!!! |
| 15:54.37 | ruied | The call must be always answered with moh.... two extensions answering with penalty 1 (the one that is available should ring), if both are busy, it should go to another twopenalty 2 extensions (to help the penalty 1 when needed). The penalty one should notice that there are entering calls. Penalty 2 should just be taking calls if bothe penalty1 users are on a call each... |
| 15:55.48 | [TK]D-Fender | There is no "notice entering calls" without them ringing as well. |
| 15:56.01 | [TK]D-Fender | (by * means) |
| 15:56.28 | [TK]D-Fender | penalty levels tend to increase the # of people included. |
| 15:56.58 | [TK]D-Fender | If you wan to cascade to a 2nd group, then you should not ring inuse and then failover to another queue |
| 15:57.18 | [TK]D-Fender | but as long as it has people to ring it shouldn't increment except based on time. |
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| 16:03.06 | ruied | hmm, ok, getting the picture... |
| 16:09.49 | ruied | It seems that the best option is to stop ringing inuse extensions. Two penaly1 users and Two penalty2 users, if both penalty1 are busy the it will try penalty2 users, meantime if one penalty1 is free than it goes again to the panalty1 priority users to pickup the calls... |
| 16:11.21 | [TK]D-Fender | Sttarting to sound reasonable |
| 16:12.00 | ruied | yes... |
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| 16:14.56 | ruied | thanks |
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| 18:22.29 | _boot | Samot: just tried SRTP on a Grandstream hardware phone and it seems to work. I guess baresip does something wrong when srtp gets involved :| |
| 18:26.10 | nny | Have a provider not working with call forward due to a diversion header. In which version of Asterisk was diversion_header=no implemented? |
| 18:26.32 | nny | er send_diversion=no |
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| 18:52.29 | adeeln | anyone live in the GTA area looking for a job doing * dialplan work? |
| 18:53.49 | [TK]D-Fender | Physical presence required? |
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| 19:06.31 | adeeln | [TK]D-Fender: possibly negotiable |
| 19:06.49 | adeeln | depends upon how reliable the candidate is |
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| 19:32.12 | _boot | argh, why does the world of voip have it in for me |
| 19:41.16 | _boot | decided to enable tls on a hardware phone, it calls outbound fine but when asterisk calls it, it fails but without any error - the last thing I get is 'SSL certificate ok' before a long wait and then a 503 Service Unavailable. has anyone else experienced this? |
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| 20:17.57 | adeeln | _boot: depending upon the phone, it might be doing mutual auth, so asterisk also needs to present a cert. if that cert doesn't have the right fields (e.g. CRL, signed by public CA, etc) then that could cause it |
| 20:18.06 | adeeln | _boot: wireshark is your friend here |
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| 20:22.59 | _boot | seems to be fixed by enabling a setting to use the ephemeral port for more than just registration or something - i guess if it happens again I'll pop out wireshark :V thanks |
| 20:23.23 | _boot | i wish the options on these horrible phone web UIs had a little more information beside them |
| 20:36.06 | adeeln | _boot: you mean on * to set the insecure=port,invite ? |
| 20:38.46 | _boot | no it was some obscure-ish option on the phone itself |
| 20:41.22 | _boot | "Use Actual Ephemeral Port in Contact with TCP/TLS" probably just sidestepping the issue here but its working for now haha |
| 20:42.45 | _boot | anyway, i'm off - later! |
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| 20:58.37 | cmendes0101 | Whats up with the spam calls for Astricon this year. Told them 3 times I'm not going |
| 20:58.56 | freebs | just go |
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| 21:00.29 | cmendes0101 | lol |
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| 23:25.05 | nny | what version was send_diversion added to asterisk? |
| 23:25.13 | nny | under sip.conf |
| 23:26.30 | WIMPy | only knows that it exists in 11. |
| 23:27.09 | nny | I may update specifically to enable it then. Level3 is not allowing us to forward calls because it is using the 4 digit extension shown in the diversion header as the CID and failing :\ |
| 23:28.01 | WIMPy | Why are you sending a 4 digits number then? |
| 23:28.32 | WIMPy | Doesn't sound like somethign that should leave your network. |
| 23:30.24 | nny | Call forward from Cisco 79XX phones |
| 23:31.09 | nny | If someone calls into the phone, and then it dials out to connect to the cfwd set, it sends the 4 digit of the cfwd phone as a diversion header |
| 23:31.13 | WIMPy | Correct it in your dialplan. |
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| 23:31.39 | nny | I am setting CallerID(num) and (name) there before the call |
| 23:32.04 | nny | oddly this is only an issue with this type of dialplan with level3 |
| 23:32.28 | nny | er yeah, the diversion header should be the callee |
| 23:32.41 | nny | maybe I can just set it there and match the channel variable for the caller |
| 23:32.44 | nny | not callee |
| 23:33.13 | WIMPy | Or use propper dialplan based forwarding. |
| 23:33.24 | nny | tell that to the users lol |
| 23:33.30 | nny | they want the phone to do it, not my bag |
| 23:34.24 | nny | I could do a * code and have the dialplan do it, but they like the phone interface for it. And they are using old Cisco phones in SIP mode which I don't usually recommend over SPA or modern SIP phones |
| 23:34.44 | WIMPy | Why do I have that feeling that it would work in a sensible way if it used SCCP? |
| 23:34.49 | nny | hahah |
| 23:35.42 | nny | or just use sensible SPA5X4G models.. or Polycomm.. or anything |
| 23:36.17 | nny | I am gonna do some more testing and whatnot, thanks |
| 23:40.08 | nny | @WIMPy how does a proper phone handle it, does the phone send the header to replace Callerid(num)? On other systems even when those are set the phone sends the proper callerid on a forward. |
| 23:43.33 | WIMPy | Forwarding is a switch/server based service. Te phone only sends a request to enable/disable it. |
| 23:44.37 | WIMPy | Terminal base diversion would be called deflection. |
| 23:56.24 | nny | WIMPy: since the phone stores the number forwarded to I assume this is deflection |
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