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| 01:48.15 | Spengler1 | is it possible to automatically initiate a command after a sip registration? |
| 01:51.15 | pabelanger | not native to asterisk |
| 01:51.32 | pabelanger | you'd have to write something that polled AMI looking for new registrations |
| 01:52.52 | Katty | can it make me coffee?! |
| 01:53.01 | [TK]D-Fender | Katty: Mine does, so yes |
| 01:53.03 | Katty | i bet i could do it with a raspberry pi |
| 01:53.14 | Spengler1 | basically i want members to know if they are in the call queue |
| 01:53.16 | [TK]D-Fender | Katty: X-10 CM11A <- |
| 01:53.23 | Spengler1 | with a lamp on the phone |
| 01:53.37 | Spengler1 | any suggestions for doing this? |
| 01:53.54 | Katty | [TK]D-Fender: oooh, fancy |
| 01:53.56 | Spengler1 | I don't want a member to forget they are in the queue and leave |
| 01:53.58 | [TK]D-Fender | Spengler1: How do you go from "sip registration" to in call queue"? |
| 01:54.05 | [TK]D-Fender | Spengler1: Those 2 are nothing alike |
| 01:55.06 | Spengler1 | if they press an alternate line then it would register and initiate the dial sequence to add itself to the queue ; i'd put an indefinite timeout on the dial plan for entering the queue so the lamp would be lit until they hangup |
| 01:55.20 | Spengler1 | my first time doing call queues so i realize that might sound stupid :) |
| 01:55.42 | [TK]D-Fender | Getting less coherent |
| 01:56.42 | [TK]D-Fender | clarify and validate how phones are even becoming deregistered in the first place... and what this call is that you make sound like is going on indefinitely. |
| 01:57.32 | Spengler1 | do you setup lamp indicators for your queue members? |
| 01:58.16 | [TK]D-Fender | I set up BLF for the PHONES they use |
| 01:58.30 | [TK]D-Fender | As well as live queue stats on the phones themselves including login status, etc |
| 01:58.39 | [TK]D-Fender | But you are mixing terms in a dangerous way |
| 01:59.14 | [TK]D-Fender | So lets rewind your scenario a few steps |
| 01:59.26 | [TK]D-Fender | And maybe we can thresh out something to fit |
| 01:59.28 | Spengler1 | Okay good idea :) |
| 02:00.05 | Spengler1 | So I want someone to enter a queue and light an indicator on the phone so they will know they are in the queue |
| 02:00.24 | [TK]D-Fender | "they" = "caller"? |
| 02:01.00 | [TK]D-Fender | Actually... you use mutiple "they"'s one of which certainly isn't |
| 02:01.01 | Spengler1 | I'm sorry wrong wording ; I want the support rep to know they are a member |
| 02:01.25 | [TK]D-Fender | Please stop abusing vague personal pronouns.... we'll report you to Miriam Webster |
| 02:01.37 | [TK]D-Fender | You want your MEMEBER to know they've logged in. |
| 02:01.39 | Spengler1 | lol sorry |
| 02:01.42 | [TK]D-Fender | Ok, that's a start |
| 02:01.42 | Spengler1 | yes that is right |
| 02:01.55 | [TK]D-Fender | What phone models are they using precisely? |
| 02:02.07 | Spengler1 | Digium D50 |
| 02:02.29 | Spengler1 | I am currently using 1 for testing |
| 02:03.21 | [TK]D-Fender | OkYou can set a custom DEVICE_STATE flag and use that as a BLF on your phone. Set it when they use your login extension, turn it off when they use your logoff one |
| 02:03.34 | [TK]D-Fender | this is all dialplan and setting the BLF on one of the buttons |
| 02:04.22 | [TK]D-Fender | I would use a combo login/logout extension for this so the BLF is a speed-dial triggered to it so it's both the indicator of the status AND the button to trigger the reversal |
| 02:04.24 | Spengler1 | thanks for your help |
| 02:04.44 | [TK]D-Fender | "Not lit? Push button to log in. Gets lit on exit" |
| 02:05.22 | Spengler1 | I see ; so I am going to have to get familiar with DEVICE_STATE |
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| 02:06.21 | [TK]D-Fender | Spengler1: https://wiki.asterisk.org/wiki/display/AST/Function_DEVICE_STATE |
| 02:07.22 | [TK]D-Fender | Spengler1: Set(DEVICE_STATE(Custom:lamp1)=BUSY) <- like this sample you invent a name of the top of your heaad (I'd use a member #), and make the "hint" associated to it map to their login/out exten |
| 02:08.12 | Spengler1 | makes sense now |
| 02:08.31 | Spengler1 | I just have to toy around with it and get it the way I want |
| 02:10.12 | [TK]D-Fender | How many members are you dealing with? |
| 02:10.33 | Spengler1 | probably 3 |
| 02:10.45 | Spengler1 | they have an old pbx now ; they come in press a button |
| 02:11.00 | [TK]D-Fender | There goes about 5 minutes of your time :) |
| 02:11.13 | Spengler1 | lol |
| 02:11.27 | Spengler1 | its pretty fun learning this stuff |
| 02:11.34 | Spengler1 | very interesting |
| 02:11.35 | [TK]D-Fender | When you learn * in a DIY approach you can do jsut about anything.... |
| 02:12.51 | [TK]D-Fender | including coffee (see above) |
| 02:12.52 | Spengler1 | I have been messing with it for years ; just never really got into queues and such ; very powerful |
| 02:13.36 | [TK]D-Fender | Certainly a practical little thing. |
| 02:14.43 | [TK]D-Fender | Digium phones have application programming you can do with them to interact with your server and (maybe?) web/XML type services in a way similar to other phones like Aastra/polycom/Cisco whereby you might be about to push more live data right to the screen of the phone itself |
| 02:15.18 | [TK]D-Fender | I have used both Polycom & Aastra phones with thier respective micro-browser interfaces where I'd feed out live queue stats every 10 seconds |
| 02:15.19 | Spengler1 | what is your opinion on the digium phones? I was told in here that not many use them |
| 02:16.11 | [TK]D-Fender | My 4 agents would see the login status of everyone, who's on a call, how many callers waiting in 2 different queues, and the VM box count for 2 boxes associated wwithe overflow from those 2 queues |
| 02:16.41 | [TK]D-Fender | They are relatively new on the market. They have a few plusses. It's a mixed bag in all honesty |
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| 02:17.25 | Spengler1 | so for overflow you send the caller into voicemail? |
| 02:17.35 | [TK]D-Fender | I'm not sure how the firmware has evolved and that was one of my bigger gripes |
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| 02:17.57 | [TK]D-Fender | yes, if a caller sits in queue too long I dump them to VM. I also do this if there are no agents logged in (afterhours) |
| 02:19.50 | Spengler1 | I was thinking about doing the same thing ; so my dial plan hits the queue , if no members are logged in it kicks them over and rings on all lines for 60 seconds then shoots into voicemail |
| 02:20.23 | [TK]D-Fender | Certainly an idea |
| 02:20.35 | [TK]D-Fender | Depends on how you wnt it to work. That's the joy of it. |
| 02:21.23 | Katty | snack time! |
| 02:21.29 | Katty | raids [TK]D-Fender's fridge. |
| 02:22.17 | [TK]D-Fender | Katty: I just stocked up. All healthy-like and everything :p |
| 02:22.37 | [TK]D-Fender | Outside of my fridge I do have an open pack of baklava however ;) |
| 02:36.24 | carrar | I'm on my way! |
| 02:36.45 | carrar | Prepare the snacks! |
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| 05:07.55 | igcewieling | Anyone else notice a significant increase in toll fraud? |
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| 05:42.30 | ChannelZ | how do you mean |
| 05:42.32 | pcAngel | I can't figure out why when I originate a call to a SIP device, to a dialplan extension that uses the Dial application, with the 'r' option, the SIP device does not play any ringing. The SIP device is a handset and is answered through speakerphone or picking up the handset. |
| 05:42.41 | pcAngel | Where should I be looking for the problem? |
| 05:43.08 | ChannelZ | most devices generate their own ringing and don't need r |
| 05:43.10 | pcAngel | Asterisk 11.2.1 and 10.7.0 |
| 05:44.00 | pcAngel | I get that, but I thought 'r' was to override that? |
| 05:44.39 | ChannelZ | no r tries to generate ringing in rare cases where the channel doesn't supply progress properly |
| 05:46.36 | pcAngel | The device rings on an outbound call through the same context, but when a channel is opened to the phone and then sent to a Dial() it doesn't. Is there a way to get around that within asterisk or do I have to change the phone settings? It's two different sip phone manufacturers, and one set isn't quite in my control |
| 05:47.42 | ChannelZ | What is the call path? |
| 05:48.08 | ChannelZ | You say 'sent to a Dial', what are you dialing? |
| 05:51.36 | pcAngel | the dial application. I am using port 5038 to send Originate with Channel: SIP/ext@remotesite, Exten: XXXXXXXXXX, Context: 200_Agent, Priority:1. XXXXXXXXXX@200_Agent does some stuff and then calls Dial(SIP/${EXTEN}@{TRUNK_IP},,TrkC) |
| 05:55.09 | pcAngel | Does that answer your question? |
| 05:55.23 | pcAngel | I haven't heard the phrase call path before |
| 05:56.47 | ChannelZ | So you're connecting two remote numbers? |
| 05:57.02 | pcAngel | Yes |
| 05:57.42 | pcAngel | some of my customers complain of that problem using our java sip client as well, which connects directly to a SIP device on the same asterisk server, however that complaint is inconsistent |
| 05:58.03 | ChannelZ | So whatever SIP/ext@remotesite is gets called, they pick up, and then you're saying they hear nothing while the other one rings (SIP/${EXTEN}@.....) |
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| 06:10.43 | pcAngel | That's correct |
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| 06:11.10 | pcAngel | in my test environment, remotesite is another asterisk server, and the extension being called at it is my sNOM 7xx |
| 06:11.41 | pcAngel | and at my clients site they are using some cisco SIP gear |
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| 07:04.46 | deo | hi guys need some help |
| 07:05.20 | deo | a log from my asterisk server says a call is running over 7 hours already |
| 07:05.22 | deo | Zap/pseudo-120596957 from-zaptel s 1 Rsrvd (None) (None) 07:05:09 (None) |
| 07:06.05 | deo | need to soft hangup this channel coz no body is using the phone now |
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| 07:06.58 | deo | executing soft hangup will only display " Requested Hangup on channel 'Zap/pseudo-1205969571'" but wont hangup the channel |
| 07:07.12 | deo | any ideas guys? thanks in advance... |
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| 07:08.55 | ChannelZ | well the psuedo channel isn't a real channel. |
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| 07:10.21 | deo | ChannelZ: what to do? |
| 07:10.28 | kaldemar | deo: nothing |
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| 07:10.48 | fabiobik_ | Hi everyone |
| 07:12.03 | fabiobik_ | my native language is not english so i will try my best to describe what i need |
| 07:12.48 | ectospasm | ~ask |
| 07:12.48 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:12.57 | fabiobik_ | so if someone call to my sip extension and its not onine, foward to my mobile |
| 07:13.33 | fabiobik_ | if rings and the sip extension not awnser, call to my mobil |
| 07:13.37 | fabiobik_ | *mobile |
| 07:14.07 | deo | kaldemar: so this will stay as it is |
| 07:14.12 | kaldemar | fabiobik_: in your dialplan, inspect the DIALSTATUS variable after you have dialed your sip phone. |
| 07:14.33 | fabiobik_ | kaldemar: can you give me a example? |
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| 07:16.25 | kaldemar | deo: as it should. |
| 07:16.38 | deo | thanks kaldemar |
| 07:19.01 | kaldemar | fabiobik_: Goto(s-${DIALSTATUS},1) or GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?mobile) or something like that. |
| 07:19.26 | kaldemar | "core show application Dial" in CLI will show you the possible status after a Dial(). |
| 07:30.57 | fabiobik_ | kaldemar: do you know how the Playback() its processed? i mean, i can awnser the call, do a playback and loop that playback untill someone accepts the call |
| 07:32.43 | ectospasm | fabiobik_: Playback() plays whatever list of files you provide. Once. To have it loop you'll need dialplan logic to tell it to jump back to the Playback() application call |
| 07:33.52 | fabiobik_ | ectospasm: do you understand what ive asked? until the extension not answer play music |
| 07:34.53 | kaldemar | fabiobik_: Playback is not used for that. |
| 07:34.55 | ectospasm | fabiobik_: I don't understand what you ask. "until the extension not answer play music" does not parse, it's missing keywords which would give complete meaning to the phrase. |
| 07:35.25 | fabiobik_ | until the extension not answer, play music |
| 07:35.26 | kaldemar | fabiobik_: either use the m() option for Dial or use a queue. |
| 07:35.48 | kaldemar | "until not answer" does not make sense. i guess you mean "until answer". |
| 07:36.19 | fabiobik_ | yes you right kaldemar |
| 07:36.26 | fabiobik_ | sorry xD |
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| 08:10.45 | deo | hi guys.. is there any debug command in asterisk to show if calls really get into another box.. |
| 08:11.23 | deo | i mean.. for iax2 connection.. it seems that when i dialled a number that is intended to another server.. i cant get through.. |
| 08:11.37 | deo | but days before, it went though.. |
| 08:11.41 | ectospasm | deo: is this SIP? "sip set debug on peer <peer>" will tell you that, but only from this Asterisk systems perspective (you should see the responses) |
| 08:12.09 | deo | ectospasm: iax2 connection between 3 servers |
| 08:12.34 | ectospasm | iax2 doesn't have as nice debugging tools |
| 08:12.43 | ectospasm | but you can use tcpdump/wireshark for that. |
| 08:12.54 | ectospasm | run tcpdump on all three servers |
| 08:13.03 | ectospasm | use wireshark to merge the three files |
| 08:13.11 | ectospasm | examine the VoIP/IAX2 flow |
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| 08:53.33 | deo | ectospasm: based on logs i found these Dial failed due to trunk reporting BUSY - giving up |
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| 08:54.04 | ectospasm | deo: that just tells you what that Asterisk system is seeing. |
| 08:54.31 | deo | running iax2 show peers on both servers displays connection is ok |
| 08:55.00 | ectospasm | deo: the tcpdump/wireshark method will show you the actual call flows, to see if the first Asterisk system is just not receiving a response, or whether the second or third systems aren't sending their responses. |
| 08:55.15 | ectospasm | deo: is this the first time you've set up this complicated configuration? |
| 08:55.20 | ectospasm | deo: has it ever worked? |
| 08:55.30 | deo | hmmnn i will try ectospasm , i didnt used wireshark before :) |
| 08:55.43 | deo | ectospasm: yeah it worked for the last few months. |
| 08:56.01 | deo | i mean this is really working.. i would have guessed that our network is the problem |
| 08:56.36 | ectospasm | well, tcpdump/wireshark should really shed some light. |
| 08:56.53 | deo | will try it ectospasm |
| 08:56.54 | deo | thanks |
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| 10:46.10 | jkroon | does it matter if the payload type numbers differ on two sides of an rtp stream (asterisk sets DTMF to 101, peer sets to 96, they claim that's why DTMF is not working). |
| 10:46.27 | Greenlight | jkroon: yes |
| 10:46.32 | Greenlight | I had to alter asterisk source |
| 10:46.40 | Greenlight | Each side blames the other and claims the other is wrong ;/ |
| 10:46.54 | Greenlight | And hi :) |
| 10:46.57 | jkroon | hi :) |
| 10:47.18 | jkroon | ok, now my jaw hurts ... the ground is hard. |
| 10:47.23 | Greenlight | Yea |
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| 10:47.49 | jkroon | ok, so if I now go and modify rtp_engine.c to match their value of 96 that'll break it for all my other SIP peerings?! |
| 10:47.53 | Greenlight | We just do it by default on installs now, if they're going through our wholesale carrier, since they use the one that Asterisk doesn't by default |
| 10:48.24 | jkroon | i'm sorry, that still sounds wrong. |
| 10:48.31 | Greenlight | I seem to rememeber we add it in, rather than replace... lemme check the diff |
| 10:48.48 | jkroon | duplicate the line to add a type 96? |
| 10:48.53 | Greenlight | Yea |
| 10:48.55 | jkroon | that still sucks. |
| 10:49.11 | Greenlight | I agree |
| 10:49.34 | jkroon | are those numbers standard in any way? |
| 10:49.36 | Greenlight | And, yes, we add it in. My guess is that it picks the first match for sending (?) |
| 10:49.52 | jkroon | surely we need to send to them using the type number they advertised, and they send to us using the type number we advertised? |
| 10:50.07 | jkroon | leifmadsen, ?? |
| 10:50.17 | Greenlight | I wasn't aware that it got advertised, if it does then that makes sense. |
| 10:50.35 | Greenlight | I remember looking up the SIP RFC at the time hoping to slap my wholesale carrier with it, but alas it wasn't too specific |
| 10:50.50 | jkroon | well, it forms part of the SDP being sent back and forth. |
| 10:51.07 | jkroon | SIP knows nothing of the SDP, you need to look at the RTP RFC. |
| 10:51.23 | Greenlight | Ahh, that makes more sense :) |
| 10:52.35 | jkroon | returns to hacking some exim stuff and being totally disgusted with RTP ... :( |
| 10:52.37 | Greenlight | Perhaps there's a more elegant solution, or a more official stance on who's right. |
| 10:52.56 | jkroon | yea, thus why i'm asking here - peeps here tend to understand these things better than me. |
| 10:53.00 | Greenlight | BUt from what I could tell it was one of the big guys like Cisco digging their feet in |
| 10:53.32 | jkroon | oh, and it's rfc2833 that defines the dtmf payload stuff. |
| 10:53.50 | Greenlight | Ahh yea I remember looking at this at the time |
| 10:54.23 | Greenlight | 96 being "dynamic payload" |
| 10:55.44 | Greenlight | Hmm... https://lists.cs.columbia.edu/pipermail/sip-implementors/2010-November/026083.html seems to suggest the payload type should be negotiated, just like the codec is |
| 10:56.22 | jkroon | this payload format does not have a static payload type number, but uses a RTP payload type number established dynamically and out-of-band. |
| 10:56.22 | jkroon | out-of-band meaning sdp? |
| 10:56.28 | *** join/#asterisk phillcz (~fjenicek@2001:718:1803:12:84b9:7ef6:e00f:55f2) |
| 10:56.46 | jkroon | Greenlight, your link says exactly what I said. |
| 10:56.52 | Greenlight | Yea |
| 10:57.44 | Greenlight | I don't even remember 96 being defined before I added it to rtp_engine.c |
| 10:57.54 | jkroon | so basically vodacom (one of the biggest voice providers in SA) is now throwing their weight around, forcing me to violate RFC specifications because the equipment they invested in is broken. |
| 10:58.06 | jkroon | it's not. |
| 10:58.17 | jkroon | but my peer is sending telephone-event with type 96 |
| 10:58.19 | Greenlight | jkroon: My wholesale carrier suggested those who use asterisk should use DTMF in band. |
| 10:58.41 | jkroon | eesh no, then i have to decode all g729 streams and *hope* to get somewhat reliable DTMF :( |
| 10:58.47 | Greenlight | Indeed ^^ |
| 10:58.56 | jkroon | SIP INFO then rather. |
| 10:59.19 | Greenlight | Yea |
| 11:00.26 | Greenlight | Either way it does seem a messy situation. |
| 11:01.33 | WIMPy | Isn't VOIP and especially SIP great? |
| 11:01.42 | Greenlight | Amen |
| 11:01.45 | jkroon | ok, but the bottom line is my standpoint that the provider is wrong is backed. |
| 11:01.55 | jkroon | WIMPy, don't you just love ISDN even more? |
| 11:02.14 | Greenlight | jkroon: Yes, that's the conclusion I came to, as they should accept 101. |
| 11:02.16 | WIMPy | jkroon: I sure do. I linke things that work. |
| 11:02.27 | Greenlight | But when ISDN does break... |
| 11:02.32 | jkroon | rofl, BRI does NOT "just work" for me :p |
| 11:02.36 | WIMPy | Or more and more "used to work" :-( |
| 11:03.02 | WIMPy | jkroon: BRI or DAHDI? |
| 11:03.10 | jkroon | :p don't know, don't care. |
| 11:03.21 | jkroon | client is lost. but they're not the only one I had issues with. |
| 11:04.33 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 11:04.33 | WIMPy | New features when using Asterisk. |
| 11:06.06 | phillcz | Hi, any asterisk guru in here? :) |
| 11:06.48 | WIMPy | Anyone use Asterisk in here? |
| 11:06.54 | WIMPy | ~ask |
| 11:06.54 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:06.59 | phillcz | A customer of ours is experiencing strange asterisk issues. He claims that sometimes callers while listening to music on hold can talk to each other. |
| 11:06.59 | phillcz | He is often transfering calls to a queue and says, that when a second call comes in and is put on hold, then the two callers are connected. It happens rarely, but happens. |
| 11:06.59 | phillcz | Unfortunatelly, I don't have access to the installation. Nothing interesting in the verbose logs, I requested more debug logs. Also tried JIRA but I couldn't find anything. |
| 11:06.59 | phillcz | Any ideas where to look? Anyone experienced something similar? |
| 11:06.59 | phillcz | Asterisk 1.18, SIP trunks |
| 11:06.59 | Greenlight | looks around |
| 11:07.28 | Greenlight | 1.18 ? |
| 11:07.36 | phillcz | 1.8.18.0 |
| 11:07.43 | Greenlight | Ahh :)( |
| 11:07.50 | WIMPy | phillcz: Are you sure he knows how to operate his phone? |
| 11:07.57 | Greenlight | My first bet as well |
| 11:08.10 | Greenlight | They've got auto-conference enabled or something silly |
| 11:08.18 | phillcz | WIMPy: I hope so, unfortunatelly he's on the other side of the planet :) |
| 11:08.20 | Greenlight | My users sometimes do that on xlite |
| 11:08.26 | jkroon | yea, i haven't yet heard of "crossed lines" on voip ... except for conferencing. |
| 11:08.57 | Greenlight | The other times is when the user they "hear" is in same room, and is picked up on their own mic, and amplified to them |
| 11:09.01 | phillcz | That's a good idea. I think some cisco phones after a fw upgrade changed the position of xfer & conference buttons |
| 11:09.06 | WIMPy | The user interfaces on SIP phones are usually not that great. |
| 11:09.11 | phillcz | I'll have to let him doublecheck that. |
| 11:09.20 | WIMPy | Or sometimes just horrible. |
| 11:09.42 | *** join/#asterisk kesselklopfer79 (~kesselklo@astaro.starface.de) |
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| 12:33.44 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:35.04 | *** join/#asterisk deo (~deo@112.198.90.38) |
| 12:35.51 | deo | hello everybody, anyone familiar with this error? Got SIP response 603 "Declined" back from |
| 12:35.58 | deo | Got SIP response 603 "Declined" back from x.x.x.x. |
| 12:36.50 | file | the device has declined |
| 12:37.00 | [TK]D-Fender | deo, sure. But the raw meaning depnds on WHY they give that to you. |
| 12:37.11 | [TK]D-Fender | deo, Maybe you should tell us more. |
| 12:37.24 | [TK]D-Fender | maybe even ... |
| 12:37.25 | [TK]D-Fender | ~pb |
| 12:37.25 | infobot | A "pastebin" is 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://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 12:37.29 | [TK]D-Fender | ^^^^ |
| 12:37.48 | deo | [TK]D-Fender: the real issue is that i cannot call to outside numbers |
| 12:37.53 | *** join/#asterisk FireAndIce (~FireAndIc@203.187.232.195) |
| 12:38.10 | deo | i trace the call logs when dialling a number, and this is what ive found Got SIP response 603 "Declined" back from |
| 12:38.19 | [TK]D-Fender | deo, show us. "sip set debug on" "core set verbose 10" |
| 12:38.26 | deo | okay |
| 12:38.55 | kaldemar | deo: it is not an error. |
| 12:39.55 | kaldemar | for example if a user hits a reject or similar button in a phone when it is ringing, the phone sends a 603 decline. |
| 12:41.11 | [TK]D-Fender | Most send back a 480 for that that I've seen |
| 12:41.29 | deo | [TK]D-Fender: http://pastie.org/7123283 |
| 12:41.45 | Greenlight | You say you're calling "outside numbers", is this not your carrier rejecting the call ? |
| 12:42.04 | [TK]D-Fender | deo, I asked for SIP DEBUG in there.... |
| 12:42.04 | deo | Greenlight: i thought that also |
| 12:42.07 | Greenlight | For example incorrect IP or not registered, or no credit |
| 12:43.00 | deo | [TK]D-Fender: i did a sip set debug on > http://pastie.org/7123294 |
| 12:43.13 | kaldemar | 480 is usually the result of DND. |
| 12:46.05 | *** join/#asterisk gonewage (~gonewage@173.161.69.173) |
| 12:46.35 | [TK]D-Fender | deo, it isn't on. What version are you running ? |
| 12:46.54 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 12:48.07 | deo | [TK]D-Fender: 1.4 |
| 12:48.18 | [TK]D-Fender | ... |
| 12:48.21 | [TK]D-Fender | gah |
| 12:48.33 | [TK]D-Fender | "sip debug on" |
| 12:49.48 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 12:50.06 | deo | [TK]D-Fender: http://pastie.org/7123404 i dont know if i did something wrong... |
| 12:51.12 | [TK]D-Fender | or just "sip set debug"? |
| 12:51.32 | [TK]D-Fender | I forget at this point |
| 12:51.38 | kaldemar | "sip debug" / "sip no debug" |
| 12:52.12 | deo | sip set debug it is... |
| 12:52.51 | deo | [TK]D-Fender: which part should i paste in? |
| 12:52.58 | [TK]D-Fender | the entire call. |
| 12:53.05 | [TK]D-Fender | We should see verbose & sip debug mixed |
| 12:55.17 | *** join/#asterisk serafie (~erin@nat/digium/x-hpicjtczmfnyieut) |
| 12:55.43 | *** part/#asterisk phillcz (~fjenicek@2001:718:1803:12:84b9:7ef6:e00f:55f2) |
| 12:56.12 | fprior | Hi all, little question about Jitter: http://pastebin.com/Rx6hphiR . I don't understand completely the description. |
| 12:56.54 | deo | [TK]D-Fender: http://pastie.org/7123468 |
| 12:59.30 | kaldemar | deo: you left out the relevant parts. |
| 13:02.59 | *** join/#asterisk vlad_starkov (~vlad_star@194.67.37.86) |
| 13:07.24 | [TK]D-Fender | And this time we see "busy" |
| 13:07.31 | [TK]D-Fender | Not the same result |
| 13:07.38 | deo | kaldemar: ? |
| 13:07.41 | deo | what part? |
| 13:08.09 | kaldemar | deo: the call. |
| 13:08.53 | deo | kaldemar: i think ive pasted also the call part |
| 13:08.56 | *** join/#asterisk melter (~Melter@2001:4930:116:0:c7e:d332:48b3:766c) |
| 13:09.26 | kaldemar | all output before the first line of your latest pastebin. |
| 13:09.30 | kaldemar | deo: you have not. |
| 13:10.26 | [TK]D-Fender | deo, We see none of the dialplan before your dial there. |
| 13:10.46 | [TK]D-Fender | deo, the first line is -- SIP/goautous-out-08ce08f0 is busy <- which says the call is already long dead (and for a different reason) and we see none of it |
| 13:11.23 | deo | [TK]D-Fender: so ill paste it again? executing the call |
| 13:11.30 | Katty | [TK]D-Fender: woot! |
| 13:11.34 | kaldemar | the first paste also said "busy", even with the 603. |
| 13:11.40 | Katty | [TK]D-Fender: i do like me a stocked fridge. |
| 13:13.40 | [TK]D-Fender | Actually.. not different yet |
| 13:13.43 | [TK]D-Fender | just misleaading |
| 13:13.53 | [TK]D-Fender | ALL of them say "busy" that way. It's meaningless |
| 13:13.59 | [TK]D-Fender | So let's jsut get the whole call.... |
| 13:14.22 | [TK]D-Fender | well... |
| 13:14.48 | [TK]D-Fender | <PROTECTED> |
| 13:14.56 | [TK]D-Fender | Yup, just misleading |
| 13:15.46 | *** join/#asterisk gonewage (~gonewage@173.161.69.173) |
| 13:15.47 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:15.52 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:16.05 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:19.41 | deo | [TK]D-Fender: im sorry, but is there any way to check this? |
| 13:20.01 | *** join/#asterisk AkkerKid (~AkkerKid@23.31.20.201) |
| 13:20.03 | [TK]D-Fender | deo, Paste. The. Complete. Call. |
| 13:20.33 | AkkerKid | anyone know how to set up a Cicso IAD 2430 to be a SIP ATA? |
| 13:21.15 | deo | [TK]D-Fender: hmmnn i think my temrinal display not all the output logs.. will find some way to paste the complete call |
| 13:21.46 | kaldemar | deo: asterisk -vvvvr | tee /tmp/ast_output.log |
| 13:21.48 | [TK]D-Fender | deo, get a bigger bigger. Putty works well |
| 13:22.34 | *** join/#asterisk bchia (~Adium@nat/digium/x-bbcsgxldnncgzoob) |
| 13:23.06 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:23.06 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:31.37 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 13:37.47 | fprior | Hi again, look at http://pastebin.com/Rx6hphiR , my question is: what's mean "enabled jitterbuffer will be used only if the sending side can create and the receiving side can not accept jitter" ? |
| 13:38.50 | Rhomber | perhaps a stupid question.. but does asterisk/SIP protocol have support to send additional information about the caller? like a display picture URL? |
| 13:39.08 | Greenlight | You can add SIP headers |
| 13:39.43 | Greenlight | fprior: For example, SIP-SIP jitterbuffer would not be used, since the endpoint can accept jitter and use it's own. For SIP-DAHDI it would be used since ISDN etc cannot accept jitter |
| 13:40.37 | Rhomber | ah cool |
| 13:40.45 | Rhomber | so is this fairly common practice then? |
| 13:40.56 | Rhomber | (i.e setting contact info via SIP headers?0 |
| 13:41.33 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 13:41.54 | fprior | Greenlight so if in my test environment AsteriskA call AsteriskB I cannot test jenable, correct ? |
| 13:43.15 | Greenlight | It depends |
| 13:43.26 | Greenlight | You can also force the jitterbuffer |
| 13:43.31 | Greenlight | jbforce iirc |
| 13:43.49 | slav3_kitten | morning everyone |
| 13:44.22 | Greenlight | Rhomber: I've never used it for sending contact info like that, but am sure others have. I sometimes use it to send account related information between my boxes. |
| 13:45.21 | Greenlight | Rhomber: It depends if you have a client that's expecing the info in a particular way |
| 13:47.16 | fprior | Greenlight which are the options of jbforce ? what's mean iirc ? |
| 13:49.05 | Rhomber | I'm developing a custom version of linphone-asterisk (https://bitbucket.org/vimtura/vimphone-android/overview) .. and wanted to keep it open-sourcey while supporting cool internal functions |
| 13:49.16 | Rhomber | so this seems like a great way to make it work both ways :) |
| 13:49.28 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-xcevipevthrjjdxa) |
| 13:49.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:49.36 | Rhomber | is there any length limit to the SIP header keys? |
| 13:50.26 | wdoekes | Rhomber: in asterisk? no.. but the total sip packet size is limited to 20k -- which should be enough for everyone ;) |
| 13:51.16 | wdoekes | Rhomber: and there were issues with certain headers (route headers were limited until recently) |
| 13:53.35 | Rhomber | ah cool, i just wanted to make something unique and descriptive like X-Vimtura-Contact-Display-Url |
| 13:53.39 | Rhomber | or something |
| 13:54.06 | Rhomber | but it's kinda cool, as i'll be able to have stuff like X-Vimtura-Queue-Name and X-Vimtura-Queue-Hold-Time too |
| 13:54.13 | Rhomber | assuming I can get to the hold time information |
| 13:54.45 | *** part/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 13:54.47 | Rhomber | and then other people can use the same headers to implement their solutions :) |
| 13:56.03 | Rhomber | lol, i guess im easily excited :) |
| 13:56.30 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:56.30 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:00.46 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 14:04.43 | fprior | Greenlight, last question. AsteriskA calls AsteriskB, jbenable, jbforce settings should be on the AsteriskB side, is correct ? |
| 14:05.41 | Greenlight | AsteriskB should use a jitterbuffer, from my understanding yes. |
| 14:19.14 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 14:20.28 | fprior | Greenlight, thanks |
| 14:21.11 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 14:22.33 | *** join/#asterisk _zoom_ (~zoom@196.1.219.122) |
| 14:23.35 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:24.34 | _zoom_ | fellas, am lookn for stable (99.9%) private gsm solution, do u know asterisk + openbts will help? |
| 14:26.37 | Greenlight | Yes, looks like asterisk + openbts would allow you to do that |
| 14:27.12 | igcewieling | Anyone else notice a significant increase in toll fraud recently? |
| 14:33.16 | _zoom_ | Greenlight: the thing is really freaking me out is availability triple 9s? |
| 14:33.39 | Greenlight | 99.9% isn't that great |
| 14:33.45 | file | Greenlight, status? |
| 14:34.13 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 14:34.26 | Greenlight | _zoom_: That's almost an hours downtime a month |
| 14:34.32 | Greenlight | file: So far, so good! :) |
| 14:34.40 | file | Greenlight, nifty |
| 14:34.43 | Greenlight | Very! |
| 14:35.18 | Greenlight | SOmetimes IT scares me.... Two characters, "/" and an "n" caused downtime for hundredes of staff |
| 14:37.28 | Greenlight | I'll give it a few weeks till I call it "fixed" though - as I've had two weeks between occurances before. BUt I'm hopeful. Plus the server has been under much heavier load last few days, without issue... hitting 350% CPU |
| 14:42.09 | *** join/#asterisk Helmut_ (1f10abe5@gateway/web/freenode/ip.31.16.171.229) |
| 14:42.16 | Helmut_ | hi there |
| 14:43.09 | *** join/#asterisk horzuh (~horza@184.95.52.210) |
| 14:45.23 | Helmut_ | Following problem: I created a little speech and music for w8 at a call forwarding. On an internal test call everything works fine. On an external call, as it is made for, the speech is provided very fast - the music afterwards sounds normal. |
| 14:45.37 | *** part/#asterisk horzuh (~horza@184.95.52.210) |
| 14:45.50 | Helmut_ | whats wrong here? |
| 14:46.59 | Greenlight | Are you using differnent codecs internally and externally? |
| 14:47.20 | Greenlight | Sounds like some sort of codec translation type problem, perhaps related to the format of the speech file |
| 14:47.59 | Rhomber | IT scares me too at times :( |
| 14:48.14 | Rhomber | as does my mouse at the moment.. not sure what's up.. it's doing random shit |
| 14:49.06 | Greenlight | Such issues are usually attributed to the device between the mouse and the chair |
| 14:49.11 | Greenlight | ^^ |
| 14:49.26 | Helmut_ | the music was imported as mp3 converted via lame and sox. the speech is recorded locally as wav, sent to server and converted via sox. I used the same parameter on both convertions. |
| 14:50.04 | Greenlight | Well, are you using different codecs internally and externally ? |
| 14:50.38 | Helmut_ | How do I get to know that |
| 14:50.40 | Helmut_ | ? |
| 14:53.51 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 14:56.22 | igcewieling | Well, THAT is rather mean. Our adtran routers are getting hacked, so we put in some ACLs, the guy comes back in and disables SIP on the box. |
| 14:57.04 | Greenlight | lol |
| 14:58.40 | *** join/#asterisk ghost75 (~trechber@dslb-088-066-172-146.pools.arcor-ip.net) |
| 15:00.46 | Rhomber | LOL |
| 15:00.48 | Rhomber | and no it's not me :( |
| 15:01.52 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:02.36 | igcewieling | They MUST be getting in via telnet or ssh since that is the only thing allowed from off-net |
| 15:02.36 | Nugget | telnet is eeeeeeevil! |
| 15:03.10 | Greenlight | You allow telnet from offsite, it's that a big no no being unencrypted |
| 15:03.23 | Rhomber | telnet?!?! |
| 15:03.42 | Rhomber | that's no good at all |
| 15:04.10 | Rhomber | disable it and change all your passwords lol |
| 15:04.55 | *** join/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 15:11.54 | igcewieling | Greenlight: the likelyhood of someone hacking a router between me and the endpoint and seeing my passwords are remote. However, we will disable telnet anyway |
| 15:13.07 | Greenlight | Never been forced/tempted to telnet in from a hotel wireless connection, if an urgent issue arrises and you're away? |
| 15:13.26 | Greenlight | Better to be safe, and disable it. That's just my opinion. |
| 15:13.57 | *** part/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 15:18.20 | igcewieling | Greenlight: then I'd use SSH 8-| |
| 15:20.16 | Greenlight | :) |
| 15:23.47 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 15:24.01 | jeffspeff | how do i send a sip redirect response back in the dialplan? |
| 15:24.15 | igcewieling | jeffspeff: nothing, it happens automatically |
| 15:25.25 | jeffspeff | something like exten=1231234567,1,SIPresponse(380,12.345.12.234) |
| 15:28.12 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:28.12 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:28.34 | igcewieling | no. Asterisk requires no dialplan stuff to make it work |
| 15:29.40 | jeffspeff | i'm not asking about what asterisk requires. i'm trying to do this via dialplan and not as an automated response. |
| 15:29.45 | wdoekes | jeffspeff: Transfer() |
| 15:30.25 | jeffspeff | transfer sends response 302 doesn't it? |
| 15:30.43 | igcewieling | I was assuming you were asking about a PHONE sending a redirect. |
| 15:30.53 | igcewieling | I've NEVER gotten Transfer to work. |
| 15:31.11 | igcewieling | why not just Dial? |
| 15:32.04 | wdoekes | igcewieling: because you stay in the path, which you may not want to |
| 15:34.04 | jeffspeff | well put wdoekes |
| 15:44.41 | *** join/#asterisk jsjc (~Adium@174.252.220.87.dynamic.jazztel.es) |
| 15:58.09 | *** join/#asterisk jsjc (~Adium@174.252.220.87.dynamic.jazztel.es) |
| 16:02.04 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::1) |
| 16:02.14 | cusco | hi |
| 16:02.47 | cusco | now, with chan_motif how does one send text messaages to gmail contacts? |
| 16:03.15 | cusco | used to be JABBERSend |
| 16:16.58 | mjordan | cusco: It hasn't changed. That functionality is provided by res_xmpp: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_JabberSend_res_xmpp |
| 16:18.17 | cusco | ow.. right.. sorry |
| 16:18.28 | cusco | I thought that was rres_jabber |
| 16:19.41 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 16:20.48 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:dd0e:7835:dbbd:6c21) |
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| 16:26.28 | drmessano | res_jabber is the old module |
| 16:26.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.8) |
| 16:26.45 | igcewieling | heh, after 18 months customer finally gave us access to the PBX. Amazing what the pbx being down will get a customer to do. |
| 16:27.09 | igcewieling | wdoekes: directmedia lets you get out of the audio path. |
| 16:27.16 | *** join/#asterisk Blue_Ice (~Blue_Ice@unaffiliated/blue-ice/x-2052838) |
| 16:27.56 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 16:27.57 | wdoekes | igcewieling: why would I resort to directmedia if I can just transfer |
| 16:28.39 | wdoekes | now I cannot restart my box without killing signaling for the calls |
| 16:29.26 | Blue_Ice | sample setup: 2 asterisk servers. I want to setup a keepalived for failover between both servers. BUT both servers have half of the inbound external lines. So IF a call comes in on the second box, while it is not the "active" one. I want it to route the call to the first one. Because if it follows the dialplan, it would end up on a SIP hint for a phone which is not connected to that specific server at that time. Is that doable? |
| 16:29.49 | Blue_Ice | probably something like wrapping a macro around the sip hint or such? |
| 16:32.21 | *** join/#asterisk evil-man (~evil-man@insider-mail.icf.org.ru) |
| 16:33.33 | [TK]D-Fender | There are no macro's with hints |
| 16:33.51 | [TK]D-Fender | they are parsed at the point of subscription at best |
| 16:34.05 | [TK]D-Fender | but no good for HA |
| 16:37.07 | igcewieling | we have all devices on all servers, which doesn't help with hints, but does with other HA issues |
| 16:41.09 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
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| 17:19.38 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:24.12 | *** join/#asterisk _Corey_ (~corey@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 17:27.20 | Spengler1 | [TK]D-Fender : do you have a dial plan that I could see where you are using device states to indicated ACD queues ; I'm having a hard time finding a good example on the net |
| 17:28.00 | *** join/#asterisk kontinuity (~kontinuit@122.166.172.217) |
| 17:28.58 | [TK]D-Fender | Spengler1, You need to be more specific |
| 17:29.23 | [TK]D-Fender | Spengler1, "indicated ACD queues" is too vague |
| 17:30.00 | Spengler1 | I want to create a dial plan that sets a blf when an agent enters the call queue |
| 17:30.11 | *** join/#asterisk voxter (~hardcore@70.36.63.61) |
| 17:30.22 | voxter | is there any way to have asterisk tell me WHICH channels are currently transcoding? |
| 17:30.35 | Spengler1 | so the agent knows he is in the acd queue waiting to receive calls |
| 17:30.46 | *** join/#asterisk navaismo (~navaismo@189.241.122.125) |
| 17:31.17 | [TK]D-Fender | Spengler1, I told you that you do this in your dialplan when you dial the extension you made that adds them in the first place |
| 17:31.40 | Spengler1 | yes ; i just would like to see how it is done |
| 17:31.42 | [TK]D-Fender | voxter, Dump them all. Match by bridgechan. compare codecs. Answer |
| 17:31.59 | lorsungcu | Spengler1: can you pb your dialplan so far? specifically the bit that logs them into the queue. |
| 17:32.13 | Spengler1 | yes |
| 17:32.14 | [TK]D-Fender | Spengler1, I already directly linked you to the page for setting the states and making the hints for them |
| 17:32.20 | [TK]D-Fender | There is no more of a sample required than that. |
| 17:32.32 | Spengler1 | I know but I am having a hard time getting this. thanks |
| 17:32.43 | voxter | [TK]D-Fender: the issue is i've found asterisk will lie - at least in this version (1.4.32) - it will say 'ulaw' on both legs even though its ACTUALLY being sent thru the g729 transcoder, because allow=g729 was before allow=ulaw in the sip peer |
| 17:32.54 | lorsungcu | ah 1.4.32 |
| 17:33.00 | lorsungcu | widely known to be the most up to date version |
| 17:33.07 | voxter | of course. :) |
| 17:33.18 | voxter | The joys of "don't upgrade production systems that aren't broken" |
| 17:33.25 | lorsungcu | sounds broken to me |
| 17:33.33 | Greenlight | Why are you here if it's not broken ? |
| 17:33.43 | voxter | why, because there are 200 active channels and I'm not sure which ones are transcoding? :P |
| 17:33.51 | voxter | Its not a problem, I'm just curious of the statistic. |
| 17:34.04 | [TK]D-Fender | Doubt that highly. Feel free to show backup for it |
| 17:34.51 | [TK]D-Fender | Spengler1, Well it's 2 lines of dialplan. Show me your actual attempt for at least half of it. |
| 17:35.01 | Spengler1 | okay ; i'm getting it right now |
| 17:35.34 | [TK]D-Fender | which should come in 2 parts. |
| 17:35.38 | [TK]D-Fender | The login. And the logout |
| 17:35.39 | lorsungcu | voxter, does the peer support g729? or is it just negotiating ulaw regardless? pb sip capture showing g729 getting negotiated? |
| 17:37.31 | voxter | lorsungcu: it'll be two peers, one has allow=g729 FIRST in the list, and allow=ulaw second.. the other will be the reverse. then it'll actually set up the call using g729, but the side that has allow=ulaw first will transcode it ulaw<->g729, even though it shows the call in g729. |
| 17:37.59 | voxter | I'm describing these details from memory from an exercise i had with this about a year ago, but thats the basic jist of it. |
| 17:38.32 | voxter | Its was a matter of codec order forcing things to be transcoded even when it wasn't entirely necessary, as it could have just negotiated properly. |
| 17:38.38 | voxter | Anyways, its not a big deal |
| 17:38.46 | Spengler1 | [TK]D-Fender : http://pastebin.com/gmg8sn0X |
| 17:39.20 | Spengler1 | [TK]D-Fender : the fourth line is where i want to activate the lamp |
| 17:40.11 | [TK]D-Fender | Spengler1, no."hint" is a PRIORITY and not a dialplan application |
| 17:40.28 | [TK]D-Fender | Spengler1, And the code sample on that page showed you how to SET it with the DEVICE_STATE function |
| 17:41.10 | [TK]D-Fender | Go read your basics for presence, and that page again |
| 17:45.19 | Spengler1 | I had this earlier: same => n,Set(DEVICE_STATE(Custom:lamp1)=BUSY) |
| 17:46.09 | Spengler1 | reading up on presence now |
| 17:46.36 | [TK]D-Fender | Spengler1, that would have been a start. Now that is a FIXED custom device state. It is not yet specific to an agent. |
| 17:46.58 | [TK]D-Fender | Spengler1, But at least part of the missing bit |
| 17:47.10 | [TK]D-Fender | Spengler1, it needs to be VARIABLE.' |
| 17:47.14 | *** join/#asterisk jsjc (~Adium@189.242.220.87.dynamic.jazztel.es) |
| 17:50.17 | Spengler1 | [TK]D-Fender : it would need to be variable because the field would apply to individual agents right? |
| 17:50.29 | [TK]D-Fender | Spengler1, correct |
| 17:55.24 | *** join/#asterisk zhando (~user@c-67-161-122-77.hsd1.wa.comcast.net) |
| 17:55.57 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 17:58.24 | zhando | hi i've been playing with pbx in a flash.. I'm finding the thing is a bit fragile when you upgrade parts of it.. when i upgraded webmin, it disappeared from the pbiaf "welcome".. i upgraded centos and the welcome doesn't come up anymore - freepbx comes up in its place.. Anyone have any links for this? |
| 17:58.59 | lorsungcu | zhando: first step is to never use piaf again |
| 17:59.10 | lorsungcu | zhando: second step is to ask in #freepbx |
| 17:59.19 | zhando | lorsungcu: I thought I would get that response... |
| 17:59.40 | zhando | lorsungcu: ok point taken.. thanks.. |
| 17:59.44 | lorsungcu | no worries |
| 18:03.32 | [TK]D-Fender | Actually... you should ask in #piaf because #freepbx is not concerned with your wemin either :) |
| 18:03.38 | [TK]D-Fender | webmin* |
| 18:04.44 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 18:05.37 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:12.37 | *** join/#asterisk Free99 (~Free99@ool-4576079f.dyn.optonline.net) |
| 18:20.17 | *** join/#asterisk _Corey_ (~corey@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 18:20.18 | Free99 | hello everyone. I'm having an issue getting kamailio and asterisk working together properly. It seems that none of my voip phones will qualify despite being able to register, so I cannot call either phone |
| 18:21.04 | Free99 | I'm behind NAT, and have setup kamailio with rtp-proxy to handle the nat issue, the NAT is not an issue based on trying the echo test successfully |
| 18:21.33 | Free99 | can't seem to figure out what's up with inter-phone calling though :-. |
| 18:30.15 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:30.16 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:31.54 | Free99 | (sigh) I've been trying to understand all this stuff for the past week. It's pretty tiring. So guys, to get to the point: |
| 18:32.06 | Free99 | I cannot keep my phones registered, so no calls go between phones |
| 18:33.39 | _Corey_ | Free99: You haven't really indicated where the phones are registering... Asterisk or Kamailio? |
| 18:40.39 | *** join/#asterisk Free99 (~Free99@ool-4576079f.dyn.optonline.net) |
| 18:40.44 | *** join/#asterisk kontinuity (~kontinuit@122.166.172.217) |
| 18:40.53 | Free99 | whoops |
| 18:42.16 | lorsungcu | Free99: >> <_Corey_> Free99: You haven't really indicated where the phones are registering... Asterisk or Kamailio? |
| 18:44.32 | Free99 | lorsungcu & corey: my bad. The phones register to kamailio, but I tried to forward the registers to asterisk. If I run 'sip show peers' on the console, I get the two phones, but their status and hostname are unknown and unspecified, respectively |
| 18:45.22 | [TK]D-Fender | Free99, So far I'm not seeing you looking at actual SIP DEBUG and seeing any registers... or call attempts... or anything |
| 18:45.31 | *** join/#asterisk _Corey_ (~corey@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 18:45.55 | [TK]D-Fender | ~pb |
| 18:45.56 | infobot | A "pastebin" is 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://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:45.57 | [TK]D-Fender | ^^^^ |
| 18:46.28 | Free99 | I'll pastebin it out. Just a sec, and thanks in advance |
| 18:50.08 | Free99 | here's a phone registering (though for some reason it gets a 401, but I can still register?): http://pastebin.com/0GFCDBtP |
| 18:50.33 | Free99 | kamailio is running on sip.mycbird.com, forwards the calls to gamma.mycbird.com |
| 18:51.46 | [TK]D-Fender | Free99, it's failing auth challenge |
| 18:53.09 | *** join/#asterisk lvlinux (~n1gg@c-50-142-165-230.hsd1.tn.comcast.net) |
| 18:54.42 | lvlinux | anybody know anything about the Avaya G11 PSTN gateway? Does it work w Asterisk? |
| 18:56.42 | Free99 | http://pastebin.com/VxqhYPmb |
| 18:57.00 | Free99 | that's me trying to call from the "auth failed phone" to the other phone |
| 18:57.14 | [TK]D-Fender | lvlinux, everything points to it running a proprietary protocol, so no.... |
| 18:57.36 | Free99 | the thing is, if auth is being rejected, how am I registering? |
| 18:57.49 | Free99 | or even able to talk to the asterisk? I turned guests off |
| 18:57.56 | [TK]D-Fender | Free99, Looking for 7968197610 in a2billing (domain sip.mycbird.com) <- call is authed.... and rejected due to a DIALPLAN ERROR |
| 18:58.49 | lvlinux | [TK]D-Fender: k thanks - figured it was proprietary but couldn't really find out for sure. |
| 18:59.57 | Free99 | hmm. |
| 19:01.19 | Free99 | [tk]d-fender: so the fact that I didn't put the other phone's extension into the dialplan is what causes it to fail? |
| 19:01.22 | lorsungcu | tm1000: you have any experience or access to a grandstream GXP2124? |
| 19:01.35 | lorsungcu | wrong channel |
| 19:02.05 | [TK]D-Fender | Free99, you don't have an exten to match the number that was dialed. |
| 19:02.21 | [TK]D-Fender | Free99, "Looking for 7968197610 in a2billing" |
| 19:02.54 | Free99 | [TK]D-Fender: but I set the system up to be realtime, so.. |
| 19:03.10 | Free99 | I'm confused about if I have to use switch or dial |
| 19:03.29 | Free99 | (I started using asterisk last week if you couldn't tell lol) |
| 19:05.04 | Free99 | [tk]D-fender: have you ever heard of a2billing? |
| 19:05.26 | [TK]D-Fender | Free99, Your context isn't right to match it. Also given the processing that a2b does there i no point in using RealTime for it at all... |
| 19:05.52 | Free99 | what do you mean? |
| 19:05.59 | [TK]D-Fender | Free99, Yes. If it a PHP LCR script for every chump who thinks he'll be the next big calling-card business :p |
| 19:06.20 | lvlinux | lol |
| 19:06.34 | [TK]D-Fender | Free99, a2b's functional dialplan requirements are all of a half-dozen fixed lines |
| 19:06.49 | [TK]D-Fender | Free99, All the logic is in their script.... so no point in having the dialplan in a DB |
| 19:07.14 | Free99 | lol that's kind of what I'm setting it up as. Oh. I just realized what you meant |
| 19:07.26 | Free99 | well I'm not even there yety |
| 19:07.28 | Free99 | *yet |
| 19:07.32 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:07.32 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:07.43 | Free99 | I just want to be sure kamailio and asterisk are working together properly |
| 19:07.57 | [TK]D-Fender | Free99, well the call is fine. Your DIALPLAN is not |
| 19:08.19 | WIMPy | Extreme S&M |
| 19:09.23 | Free99 | [tk]D-fender, is there a way for me to make the extension exist automatically based on the database having a user? |
| 19:09.56 | [TK]D-Fender | Free99, that is the NUMBEr they are dialing.... you are supporsed to pass it ON to a2b to dial out. |
| 19:10.06 | [TK]D-Fender | You should have a PATTERN to match this for processing per it's table |
| 19:10.14 | [TK]D-Fender | there is nothing to 'generate' for this |
| 19:10.37 | [TK]D-Fender | You don't have to add every phone number in the world to the DB for this. |
| 19:12.00 | Free99 | [tk]: I got it, the thing is I'd just like to make it work for a test |
| 19:12.11 | Free99 | I don't even care about a2b yet, let's pretend it doesn't exist |
| 19:12.37 | Free99 | if I just had a database with some oddly titled columns, how would I pull the extensions out? |
| 19:12.39 | [TK]D-Fender | Free99, fix your dialplan |
| 19:17.18 | Free99 | [tk]D-fender: I don't want to ask you to solve this for me. But, is it impossible to get the extensions out of the same DB holding the user's VoIP info? |
| 19:17.36 | Free99 | I pretty much followed this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb |
| 19:19.37 | [TK]D-Fender | Free99, EXTENSIONS.CONF <------------- DIALPLAN |
| 19:20.01 | *** join/#asterisk BCS-Satori (~BCS-Sator@2001:470:e169:860:891:5075:f458:805c) |
| 19:20.57 | Greenlight | I've just had a problem with a switch dieing at the datacentre I've got an asterisk box in that is a billing gateway. I've realised that it's still got channels hanging around from when the network went down (missed the BYE). I don't use directemdia. What's the best solution to ensure that asterisk kills off channels that die like this; rtptimeout or SIP session timers, or both? |
| 19:22.19 | WIMPy | rtptimeout can be faster, but only works when you are in the media path, obviousely. |
| 19:23.00 | anonymouz666 | rtptimeout works for one side only? |
| 19:23.15 | anonymouz666 | ahh I thinks it's both |
| 19:23.27 | BCS-Satori | I am having problems with calling outbound on a new analog fxo card using the latest dahdi. I have six analog phone lines conneced and I am able to call inbound on all of the without a problem however I am only able to call outbound on one of the six lines. The other five lines produce a fast busy signal; however when I connect a buttset to the lines I am able to call out without issue. All lines show 52-54 Volts. Any ide |
| 19:23.27 | BCS-Satori | as where I need to begin to look? |
| 19:23.38 | WIMPy | For the receiving side of both channels. |
| 19:23.46 | WIMPy | (if both channels are SIP) |
| 19:23.57 | igcewieling | anonymouz666: only works for NON-reinvited audio |
| 19:24.33 | anonymouz666 | I remember when the telco stuck a call sending RTP frames (busy tone)... there was just one side receiving |
| 19:25.08 | Free99 | [tk] exten => _1XX,1,Dial(SIP/${EXTEN}) |
| 19:25.16 | anonymouz666 | 12 hours call duration he he. |
| 19:25.21 | Greenlight | What about SIP session timers? |
| 19:25.26 | WIMPy | There is an extra holdtimeout or so IIRC. |
| 19:25.34 | anonymouz666 | that's why it's always a good ideia to set an arbitry absolute timeout |
| 19:25.38 | Greenlight | Ideally I don't want to *have* to remain in media path |
| 19:25.46 | anonymouz666 | yes, there is hold time |
| 19:25.59 | anonymouz666 | arbitrary |
| 19:26.33 | mjordan | SIP session timers would also do the trick. What version are you running? |
| 19:26.40 | Greenlight | 11.3 |
| 19:27.08 | mjordan | k. They received a lot of TLC and should work fine in recent versions |
| 19:27.09 | Greenlight | Oh, actually, that box is 11.0.1 |
| 19:27.20 | mjordan | pretty sure it went into the initial version of 11 :-) |
| 19:27.23 | Greenlight | Cool |
| 19:27.29 | Greenlight | What about carrier support? |
| 19:27.42 | mjordan | that's a whole different question, unfortunately. |
| 19:27.45 | Katty | carrar: wilfred is done! |
| 19:27.52 | Katty | carrar: http://42ndknitstreet.blogspot.com/2013/03/wilfred-giraffe-for-henry.html <- cute bits at the bottom. |
| 19:28.07 | Greenlight | Heh - I read it'll do RE-INVITES or something if other side dones't support ? |
| 19:28.12 | Free99 | [tk]D-fender: I put this into extensions.conf "exten => _XXXXXXXXXX,1,Dial(sip/${EXTEN})" because I have 10 digit account codes. That's where I've been stuck since midday |
| 19:29.21 | WIMPy | Greenlight: With some peers calls drop randomly at session timer intervals. |
| 19:32.23 | Greenlight | I don't want that |
| 19:32.52 | Greenlight | My peers will always be either my wholesale carrier, or other asterisk 11 boxes |
| 19:33.14 | WIMPy | Should be safe |
| 19:35.35 | carrar | woah |
| 19:35.46 | carrar | is that a yellow horse? |
| 19:35.55 | Free99 | I keep getting a "cause 20 - subscriber absent" when I try to make calls between sip phones |
| 19:35.58 | carrar | err girraffe |
| 19:36.26 | Katty | carrar: yes. |
| 19:36.34 | Katty | carrar: hopefully henry won't chew his ears off. |
| 19:36.35 | carrar | does he talk? |
| 19:36.45 | Katty | no. he's just full of fiberfill. |
| 19:36.51 | Katty | but you can pretend! |
| 19:37.06 | carrar | is that Giraffe building a computer? |
| 19:37.10 | carrar | holycow |
| 19:37.12 | Katty | yes ^____^ |
| 19:37.28 | carrar | That Giraffe looks FUN!!!! |
| 19:37.36 | Katty | i'm certainly hoping so |
| 19:37.57 | [TK]D-Fender | Free99, fuirst you aren't using a2b there at all. Next I have no idea what context that line refers to (which clearly isn't [a2billing] like is being requested), and you should be USING a2billing to dial out, not just dial out to some random SIP peer. |
| 19:38.09 | Katty | carrar: which one is your favorite image of Wilfred? |
| 19:39.04 | Free99 | [tk] the thing is, why am I going to try to make a system that handles money if I don't know how to use it normally first? That line got added into a context [a2billing] so that it wouldn't complain |
| 19:39.13 | carrar | I think looking at the dog |
| 19:39.30 | Free99 | normally being "not tied to a2billing" |
| 19:39.46 | Katty | carrar: that seems to be everyone's favorite. |
| 19:40.07 | [TK]D-Fender | Free99, well your call is looking in [a2billing] and I have no proof the context exists.... and I know if doesn't have an exten to match the number that was dialed regardless |
| 19:40.44 | [TK]D-Fender | Free99, go validate the context and the extens in it |
| 19:41.04 | Free99 | [tk] that's the issue I'm trying to deal with, the lack of exten. Although I'm setting this up for a company, and getting paid for it, I'd actually like to know how asterisk works, you know? |
| 19:41.16 | igcewieling | Free99: and that is what [TK]D-Fender is helping you with. |
| 19:41.38 | igcewieling | for the likely the call is going to a DIFFERENT context than where the extension is. |
| 19:42.30 | Free99 | I mean I stuck "a2billing" as the context for the realtime users in the database. If I get rid of a2billing, or comment it out, I get a big ol' warning in the console |
| 19:42.47 | [TK]D-Fender | Free99, Your caller HAS a SIP peer that it matched. That peer specifies a CONTEXT that it's calls should land in. * looks in that CONTEXT for an EXTEN to match the number they dialed. You do not have a match in EXTENSIONS.CONF in the CONTEXT you told it to look in for the NUMBEr they are looking for. |
| 19:44.12 | Greenlight | How till sip session timers or rtptimeout effect SIP channels already active, if I made the change now... would it clear out those stale channels ? |
| 19:44.19 | Greenlight | *How will |
| 19:44.52 | [TK]D-Fender | Free99, http://pastebin.com/s4yTajHC |
| 19:45.03 | Free99 | [tk]d-fender: how do you know it matched the peer, and where is the * you are looking at? |
| 19:45.16 | [TK]D-Fender | Free99, If I tell you to look in the TOP shelf for the forks and they are really in the MIDDLE shelf ... FAIL |
| 19:45.30 | [TK]D-Fender | Free99, Found peer '8789320397' for '8789320397' from 176.58.108.118:5060 |
| 19:45.31 | [TK]D-Fender | ^ |
| 19:45.34 | [TK]D-Fender | FOUND PEER |
| 19:45.36 | [TK]D-Fender | in the big print |
| 19:45.50 | [TK]D-Fender | Free99, Looking for 7968197610 in a2billing (domain sip.mycbird.com) <--- WHAT THEY ARE looking for |
| 19:45.53 | [TK]D-Fender | AND where |
| 19:46.08 | [TK]D-Fender | darn caps inversion |
| 19:46.22 | Free99 | hey man, I appreciate the help, but I'm sorry I'm making you mad (apparently) |
| 19:46.35 | [TK]D-Fender | no, just highlighting the important terms |
| 19:47.00 | [TK]D-Fender | Free99, You have no match for "7968197610" in [a2billing] just like it says |
| 19:47.04 | [TK]D-Fender | Free99, Go fix that |
| 19:47.18 | [TK]D-Fender | Free99, make sure you have an exten there to match that number if you expect it to accept it |
| 19:49.09 | Free99 | so I'd do something like exten => _7968197610,1,Dial(sip/7968197610) ? |
| 19:49.38 | [TK]D-Fender | Free99, I would use a PATTERN instead of a fixed # for this so far.. |
| 19:49.52 | [TK]D-Fender | you seem to want to take an identical kind of action for any kind of number like that |
| 19:50.08 | [TK]D-Fender | Free99, Which is what you showed .... in one line you gave us. |
| 19:50.19 | [TK]D-Fender | <Free99> [tk]D-fender: I put this into extensions.conf "exten => _XXXXXXXXXX,1,Dial(sip/${EXTEN})" because I have 10 digit account codes. That's where I've been stuck since midday |
| 19:50.24 | [TK]D-Fender | ^ pattern |
| 19:52.46 | Free99 | (facepalm) the thing is, what is ${EXTEN} being replaced by? |
| 19:52.55 | Free99 | in this instance, or in general? |
| 19:52.56 | [TK]D-Fender | Free99, the number you dialed |
| 19:53.07 | [TK]D-Fender | (actually the EXTEN you are on... |
| 19:53.18 | [TK]D-Fender | which because of the pattern... happens to be one and the same in that case |
| 19:54.00 | Free99 | wait a sec, you mean to say that any number I dial matching the XXXX part will wind up making asterisk try to call the phone I am currently calling from? |
| 19:54.13 | [TK]D-Fender | "exten => _XXXXXXXXXX,1,Dial(sip/${EXTEN})" <--- matches a 10-digit number. And the value of ${EXTEN} will be the number itself as that's the EXTEN you are on. |
| 19:54.29 | [TK]D-Fender | not "from".... |
| 19:54.33 | [TK]D-Fender | "TO" <- |
| 19:55.56 | Free99 | that's the pattern I've been trying to use to call the other phone, but it just never works: http://pastebin.com/jhYXe8hT |
| 19:55.58 | Free99 | just tried it now |
| 19:57.36 | [TK]D-Fender | Free99, Good, it's accepting your call.... and processing dialplan! |
| 19:57.57 | Free99 | but that's where I was stuck at when I first came here! |
| 19:58.01 | [TK]D-Fender | Free99, and you don't have a SIP PEER registered with an IP that * is capable of calling so the dial fails. |
| 19:58.05 | [TK]D-Fender | Free99, Progress! |
| 19:59.41 | Free99 | you know, too many years of dealing with wildcards in shell scripts, I suddenly just realized that earlier in the conversation when you were saying "*" you meant shorthand for "asterisk" ;) |
| 19:59.50 | [TK]D-Fender | Free99, "Dial Fred!" , "I don't know where Fred is!" (dies) |
| 20:00.14 | [TK]D-Fender | Yes, * = asterisk |
| 20:01.19 | Free99 | with regards to the lack of ip... any idea why that could be? |
| 20:02.36 | [TK]D-Fender | It isn't registered <- |
| 20:02.56 | [TK]D-Fender | This looks like your previous challenge failure coming back to haunt you |
| 20:06.41 | Free99 | [tk] I poked around in mysql, it looks like neither phone has an IP associated with it |
| 20:07.20 | [TK]D-Fender | which would be the problem. |
| 20:07.31 | [TK]D-Fender | Wirth them not responding to the 401 challenge |
| 20:08.37 | Free99 | hmm. so why is that? Could it be kamailio not properly forwarding the required info? |
| 20:09.57 | *** join/#asterisk jsjc (~Adium@189.242.220.87.dynamic.jazztel.es) |
| 20:10.14 | [TK]D-Fender | Quite possible |
| 20:10.22 | [TK]D-Fender | Perhaps you should try a device more direct |
| 20:17.53 | Free99 | would tcpdump be able to show me the contents of the register requests from kamailio? |
| 20:18.26 | [TK]D-Fender | Free99, we already saw the * SIP debug |
| 20:18.41 | [TK]D-Fender | comes in. Gets challenged, Kamalio never answers back |
| 20:20.28 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 20:24.36 | Free99 | what's up with that? it comes in, gets a 401? Does that mean the password is wrong or something? |
| 20:24.44 | Free99 | password on the client, that is |
| 20:26.02 | [TK]D-Fender | it is a CHALLENGE |
| 20:26.12 | [TK]D-Fender | Your device should come back with an answer to the challenge. |
| 20:26.18 | [TK]D-Fender | It isn't an outright refusal |
| 20:26.34 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 20:27.42 | Free99 | hmm, maybe I should set port and invite as insecure? |
| 20:29.20 | igcewieling | or maybe the response is being sent to the wrong IP (nat issue)? |
| 20:32.22 | Free99 | the thing I don't get is, http://pastebin.com/VxqhYPmb <- line 42 says 401, but line 39 said peer was found? |
| 20:35.17 | [TK]D-Fender | Free99, If found an entry matching WHO they claimed to be. It CHALLENEGED them. It RESPONDED to the challenge. The challenge was ACCEPTED |
| 20:35.40 | [TK]D-Fender | 59 ACK of challenge |
| 20:35.49 | [TK]D-Fender | 72 Re-try with auth |
| 20:36.10 | [TK]D-Fender | 109 begin call processing check for dialplan |
| 20:36.26 | [TK]D-Fender | 128 Wish I had a dialplan match |
| 20:37.25 | igcewieling | LOL! My boss is great, he sent this to someone " Did you just lump together TT's on 4 different issues, over 2 different types of services and a Add/move/change thrown in? Then to top it all off, send it to IPMAX, which isn't applicable to this client in any way shape or form? " |
| 20:37.54 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 20:38.01 | [TK]D-Fender | Checkout time here, heading home, BBL |
| 20:38.05 | *** join/#asterisk apb1963__ (~apb1963@174.134.117.244) |
| 20:42.41 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.smb.curriegrad2004.ca) |
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| 21:00.05 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:00.54 | Rhomber | does anyone know if the default incoming call screen can be used in an app? |
| 21:00.57 | Rhomber | for like voip calls? |
| 21:18.01 | [TK]D-Fender | What "default incoming call screen" ? |
| 21:20.12 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-052-224.pools.arcor-ip.net) |
| 21:23.04 | lorsungcu | for like voip calls, TK |
| 21:23.15 | [TK]D-Fender | What "default incoming call screen" ? |
| 21:23.41 | lorsungcu | for like voip calls. |
| 21:26.04 | [TK]D-Fender | Ok, one more time, a little more vague please |
| 21:26.58 | lorsungcu | not sure how much more precise i can be. |
| 21:27.22 | lorsungcu | you've got voip calls. |
| 21:27.24 | lorsungcu | and a screen |
| 21:27.27 | lorsungcu | so for like them |
| 21:27.31 | [TK]D-Fender | describing what the "screening" actually IS. |
| 21:27.44 | lorsungcu | yeah i dont know about that. bit over my head. |
| 21:27.46 | [TK]D-Fender | and how it's "default" as though that meant you could change it |
| 21:28.56 | lorsungcu | Rhomber does that describe your issue? |
| 21:29.02 | lorsungcu | with the screens and the defaulting and whatnot |
| 21:31.01 | igcewieling | lorsungcu: VoIP calls don't have screens. |
| 21:31.14 | lorsungcu | dammit |
| 21:31.14 | ChannelZ | This feels like a "Dude Where's My Car" scene |
| 21:31.19 | igcewieling | However, individual phones may or may not have a screen. without knowing the EXACT MAKE AND MODEL of the phone we cannot help you. |
| 21:32.15 | igcewieling | ChannelZ: I can only assume english is not lorsungcu's primary language |
| 21:32.48 | lorsungcu | igcewieling: <Rhomber> does anyone know if the default incoming call screen can be used in an app? |
| 21:32.48 | lorsungcu | <Rhomber> for like voip calls? |
| 21:33.05 | igcewieling | lorsungcu: no, it cannot. |
| 21:33.21 | [TK]D-Fender | Thre is no screen |
| 21:33.27 | igcewieling | If you well me exactly what device has the screen you are referring to then my answer may change. |
| 21:33.30 | lorsungcu | christ this is for Rhomber :D |
| 21:33.36 | [TK]D-Fender | Since you can't describe what it is, then it doesn't exist |
| 21:33.51 | lorsungcu | see if i ever troll you guys, ever |
| 21:33.55 | igcewieling | A Polycom phone? A digium phone? A cell phone with a SIP client? A Cisco phone? A Linksys phone? |
| 21:34.03 | ChannelZ | I don't know if he means a display screen or something in the "call screening" sense, some routine. |
| 21:34.12 | igcewieling | lorsungcu: don't worry, you are now on my do-not-help list. |
| 21:34.19 | lorsungcu | rofl |
| 21:34.55 | ChannelZ | But anyway he hasn't replied for half an hour so who knows |
| 21:50.38 | Rhomber | this was meant to go in the #android channel |
| 21:50.45 | Rhomber | but thanks for being rude. |
| 21:51.22 | Rhomber | and sorry, i was busy fixing a bug. |
| 21:51.34 | navaismo | ;) |
| 21:57.00 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 21:57.02 | [sr] | hi WIMPy |
| 21:57.13 | WIMPy | Hi [sr] |
| 21:57.26 | [sr] | WIMPy: got a question for you |
| 21:58.41 | [sr] | WIMPy: i have some instalations all with PtMP, and some work OK with two PBX's connected, the old tradicional PBX and asterisk, no noise os any other problem, in some others, few, there's alot of noise, what could be the reason do you have any idea? |
| 21:59.11 | WIMPy | A crossed cable? |
| 21:59.56 | WIMPy | Or wrong termination. |
| 22:00.04 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/7 |
| 22:00.59 | [sr] | both are connected with strait cables |
| 22:01.14 | Rhomber | i didn't realise that a question in the wrong channel could cause so much commosion :) |
| 22:02.14 | [sr] | WIMPy: one particular place, i have alot of this that never had: [2013-03-26 21:37:20] NOTICE[18074]: chan_dahdi.c:3197 my_handle_dchan_exception: PRI got event: HDLC Bad FCS (8) on D-channel of span 2 |
| 22:02.17 | jmetro | what with the defaulting and the call screens? |
| 22:02.28 | [TK]D-Fender | Rhomber: Don't worry, we're rooted out and branded the troll :) |
| 22:02.44 | [sr] | and both NTBA's (they're two) are having channels down and up after 5 seconds about every two hours |
| 22:02.44 | [TK]D-Fender | Rhomber: And are fetching hotter irons... |
| 22:02.53 | Rhomber | LOL |
| 22:03.45 | Rhomber | goes for a pee break and to get more cask wine :P |
| 22:06.04 | [sr] | WIMPy: cables are correct, well, maybe the dipswitch's on the NTBA will be the solution :) |
| 22:06.37 | WIMPy | Quite possible. |
| 22:06.39 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-221-178.pools.arcor-ip.net) |
| 22:07.00 | WIMPy | If both PBXs are terminates. |
| 22:07.22 | WIMPy | d |
| 22:10.32 | [sr] | WIMPy: i'll dipswitch it |
| 22:12.06 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
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| 22:52.49 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 23:15.01 | igcewieling | [sr]: there is a chan_dahdi.conf related to channel restarts. Disable it. |
| 23:15.19 | igcewieling | chan_dahdi.conf option related to |
| 23:16.15 | WIMPy | wonders if that should actually work on ptmp. |
| 23:18.12 | *** join/#asterisk serafie (~erin@24.214.158.242) |
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| 23:39.05 | jeffspeff | using Transfer() sends a sip 302 response. when that is sent, does it take that asterisk box out of the call path? |
| 23:48.59 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:49.54 | phix | I suppose that depends if the Transfer() redirects to the same asterisk box :) |
| 23:50.34 | jeffspeff | lol, it would obviously be a different box |
| 23:51.00 | jeffspeff | is ther any way to send a different sip response other than 302? |
| 23:52.59 | phix | ummm well a 302 is a redirect though, what other response were you after? |
| 23:58.50 | jeffspeff | i think 380 will do what i want |
| 23:59.37 | jeffspeff | i have calls coming in from my provider to a certain box and some of them i want to redirect to different boxes but take the first box out of the path |