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| 05:53.53 | D30 | D30: testing |
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| 06:24.42 | ChannelZ | D30: syntax error |
| 06:25.06 | D30 | ChannelZ: sorry i know |
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| 09:30.24 | mmoucka | hi everyone I have question. I'm working on web interface for asterisk as my school project. And if try use realtime ps_transports it shows this "Realtime mapping for 'ps_transports' found to engine 'myslq', but the engine is not available" is it possible or I have to you trasports as static realtime? |
| 09:34.31 | mmoucka | nvm I see the problem, misspelled mysql. |
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| 09:35.15 | skrusty | :] |
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| 09:57.22 | bibz | my peers are becoming "UNREACHABLE" at random times.. I have primarly Siemens Gigaset devices in use.. after a few seconds they become reachable again.. what could that be? |
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| 10:48.30 | bulkorok | shiddy gigasets |
| 10:48.49 | bulkorok | you could try with lower reg-expire time |
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| 11:31.22 | giany | hello, some devices apparently perform unhold by simply issuing a reinvite on the dialog without an SDP..this makes the call not getting out from hold..is there any workaround regarding this? |
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| 11:59.25 | bibz | I'm looking for a freelancer.. I have problems with my asterisk.. lots of "NO ANSWER" and sometimes even "BUSY" are popping up in my database.. please contact me, if you can help. |
| 11:59.54 | WIMPy | hands bibz a whip |
| 12:00.10 | RPerre | Hi. I'm building a Asterisk app using ARI. I have a Stasis application running. Outside of this app, I use ARI to originate to A. When stasis get this channel, inside de stasis i originate to B and create a bridge between the two. I get no sound between them, can anyone help me understand why? |
| 12:00.56 | bibz | a whip? what you mean by that? |
| 12:01.28 | WIMPy | bibz: To motivate people to actually answer the phone instead of ignoring it. |
| 12:02.56 | bibz | well, there are lots of "NOANSWER" with "duration" from 0-5 secs.. which I can't explain myself why it happens.. |
| 12:03.02 | skrusty | welcome back RPerre |
| 12:03.06 | skrusty | still the same issue? |
| 12:03.30 | RPerre | Hi skrusty, yes same issue :( |
| 12:03.32 | skrusty | RPerre, have you answered the first channel? |
| 12:03.42 | WIMPy | bibz: A lack of patienc from the caller? |
| 12:04.38 | bibz | WIMPy: it happens to 30-40% of all callers daily. |
| 12:05.03 | RPerre | skrusty, yes. |
| 12:05.45 | skrusty | out of interest, once bridged, does audio work if you play it specifically to the channel and not the brdige? |
| 12:06.03 | RPerre | i'll try that, give me a sec |
| 12:12.22 | RPerre | skrusty. ok, if playback to the first channel the sound passes tru. if i playback to BOTH channels, the sound passes tru. if i playback *just to the second* channel, it doesnet |
| 12:16.27 | RPerre | this make no sense.. |
| 12:16.31 | RPerre | makes* |
| 12:16.46 | RPerre | (sorry for the bad english tho) |
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| 12:24.41 | RPerre | skrusty, any ideas? |
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| 12:27.24 | skrusty | sorry, was on the phone |
| 12:27.25 | skrusty | reading up |
| 12:27.51 | skrusty | when you say *both* channels, do you mean the brdige? |
| 12:28.12 | skrusty | as play only acts on one channel |
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| 12:32.02 | RPerre | no, i mean playback in chan1, then playback in chan2 |
| 12:32.09 | RPerre | this works |
| 12:32.23 | RPerre | if i playback just to chan2, it does not work |
| 12:32.49 | WIMPy | So you're saying you can play to chan2 *after* you played to chan1? |
| 12:35.12 | RPerre | yes |
| 12:35.48 | skrusty | wtf :) |
| 12:36.09 | skrusty | what framework are you using? |
| 12:36.13 | skrusty | for ari |
| 12:36.18 | WIMPy | <skrusty> RPerre, have you answered the first channel? |
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| 12:46.38 | RPerre | WIMPy i did |
| 12:46.49 | RPerre | skrusty, phpari |
| 12:47.23 | RPerre | (sorry for my delay, i'm getting alot of phonecalls) |
| 12:47.46 | WIMPy | Phones are evil. |
| 12:47.52 | RPerre | indeed xD |
| 12:48.15 | WIMPy | And we know that you're not working for bibz :-) |
| 12:48.47 | RPerre | no, i'm not :) |
| 12:51.13 | skrusty | one thought, use a rest tool like postman to create the originate and bridge, take it down to basics. If it's still a problem, i suggest maybe there's some underlying issue |
| 12:51.36 | skrusty | if you do the rest calls yourself, you can put them in a pastebin so we can varify them and replicate! |
| 12:51.45 | skrusty | (to a degree) |
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| 12:55.48 | RPerre | i've solved it |
| 12:55.53 | skrusty | go on? |
| 12:55.59 | RPerre | the problem is not on asterisk nor ari |
| 12:56.05 | skrusty | .... |
| 12:56.08 | RPerre | i have a TELCO mediagateway |
| 12:56.17 | RPerre | wich has a firewall bug |
| 12:56.23 | skrusty | heh :) |
| 12:56.38 | RPerre | if i dont pass it a RTP packet |
| 12:56.50 | RPerre | it does not iniciate RTP session |
| 12:57.07 | RPerre | so, this playback on the first channel does the trick |
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| 12:57.19 | RPerre | i have to passit a empty sound |
| 12:57.26 | RPerre | so the channel has RTP |
| 12:57.33 | RPerre | i'm so happy right now |
| 12:57.46 | RPerre | skrusty and WIMPy, thanks for your help |
| 12:57.52 | RPerre | love you guys |
| 12:58.13 | skrusty | you're welcome :) |
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| 13:08.16 | cbeyerlein | Heya, I need some advise on PJSIP behaviour: Call comes in via Endpoint EP1, Asterisk sends code 100 to EP1. Dialplan does "Dial(PJSIP/EP2) and Asterisk instantly sends 180 Ringing to EP1, even before sending out INVITE to EP2... |
| 13:08.42 | cbeyerlein | this doesnt look right to me, call to EP2 could be rejected, or timeout or whatever, but EP1 sees "Ringing" |
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| 14:07.36 | CyberBob | Hi All. |
| 14:08.24 | CyberBob | I'm using ISDN PRI line connected with Digium TE133 card but system is not detecting hangup if the other party hangup the call. |
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| 15:34.07 | wasanzy | hi |
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| 15:39.50 | wasanzy | am still not getting OBDs working. I cat this error when I copy the call file to outgoing http://pastebin.com/HVGjhRFj |
| 15:39.56 | *** join/#asterisk newtonr (newtonr@nat/digium/x-wurskokzzwhgrfvt) |
| 15:39.56 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:42.09 | [TK]D-Fender | wasanzy: [2015-01-27 15:36:44.889] WARNING[14016]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 484019823df9bba404a4d34869538801@10.71.173.187:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 15:42.18 | [TK]D-Fender | wasanzy: And you aren't looking at the SIP DEBUG for your call attempt |
| 15:42.59 | [TK]D-Fender | wasanzy: And we clearly see stuff isn't getting responded to there. |
| 15:43.15 | wasanzy | let me get the sip debug paste |
| 15:46.44 | wasanzy | http://pastebin.com/iWQ3ppKy |
| 15:47.12 | wasanzy | thst is the sip debug |
| 15:48.19 | *** join/#asterisk Raden (~Raden@97-83-139-156.dhcp.eucl.wi.charter.com) |
| 15:48.32 | Raden | runs and jumps and gives Katty a huge hug |
| 15:48.41 | Katty | ! |
| 15:48.45 | Katty | hugskwishes Raden |
| 15:48.51 | Raden | How ya been ? |
| 15:48.56 | Katty | good :> |
| 15:48.59 | Katty | how're you? |
| 15:49.12 | Raden | busy as a bee |
| 15:50.05 | Raden | how have u been my dear ? |
| 15:50.43 | wasanzy | [TK]D-Fender: http://pastebin.com/iWQ3ppKy |
| 15:50.50 | [TK]D-Fender | wasanzy: You're sending out invites and not getting any answer from the other side... |
| 15:50.57 | [TK]D-Fender | wasanzy: Go look at what you have it pointed to., |
| 15:51.43 | wasanzy | do I really need an invite? |
| 15:53.10 | [TK]D-Fender | wasanzy: .... that's your CALL |
| 15:53.37 | [TK]D-Fender | - Attempting call on SIP/+2349096852432@outgoing for start@obd_promo:1 (Retry 1) <--------------- |
| 15:53.47 | wasanzy | insecure = invite |
| 15:53.50 | [TK]D-Fender | NO |
| 15:54.01 | [TK]D-Fender | INVITE is the pactet that starts a SIP CALL |
| 15:54.06 | [TK]D-Fender | You are doing a SIP CALL there..... |
| 15:54.26 | wasanzy | yes using call file |
| 15:54.42 | [TK]D-Fender | Yes, and that call you are making is failing |
| 15:54.45 | [TK]D-Fender | The otehr side is not answering |
| 15:54.54 | [TK]D-Fender | And by that I mean NO response at all |
| 15:55.08 | [TK]D-Fender | Not "I'm busy", or "I don't like this request". NOTHING |
| 15:55.30 | [TK]D-Fender | Go look at where it is sending the call, verify netwoking, software, etc |
| 15:55.32 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:8219:34ff:fecf:17f0) |
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| 15:56.36 | wasanzy | ok |
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| 16:03.25 | cbeyerlein | someone an idea how to prevent Asterisk/PJSIP from transmitting 180 Ringing even before sending out INVITE? http://pastebin.com/HZfjQCjb |
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| 16:12.59 | *** part/#asterisk marceloamorim (~marcelo@189-90-194-37.isimples.com.br) |
| 16:16.32 | newtonr | cbeyerlein, what about just answering the calling channel? |
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| 16:19.53 | cbeyerlein | newtonr: I can't.. we are building something like a Class4 switch, so I must report the call-results from the B-leg back to the A-Leg... |
| 16:20.18 | *** join/#asterisk marceloamorim (~marcelo@189-90-194-37.isimples.com.br) |
| 16:20.20 | cbeyerlein | so if B-leg is busy, A-leg should receive Busy too. and not indicate Ringing that is not actually happening on Destination of B leg |
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| 16:21.09 | cbeyerlein | also messes up call statistics seen on A-side like "average ringing duration" |
| 16:22.09 | [TK]D-Fender | PJ shouldn't be doing that... chan_sip wouldn't in the same scenario... |
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| 16:25.00 | wasanzy | [TK]D-Fender: I have taken a tcp dump of the request, can I send it to you for assistance? |
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| 16:25.40 | [TK]D-Fender | wasanzy: We alrady see the SIP debug.. what more is there to see? |
| 16:25.47 | [TK]D-Fender | wasanzy: Request goes out ... never comes back. |
| 16:25.52 | cbeyerlein | [TK]D-Fender: so it could be a bug? How would I properly report this? via issues.asterisk.org? Or better have a discussion first on e.g. asterisk-users? |
| 16:27.03 | *** join/#asterisk vassilux (~vassilux@mail.esifrance.net) |
| 16:27.04 | [TK]D-Fender | cbeyerlein: Could be. As this is a Digium supplied moduleI'd say you should be able to take it up on their support forums, etc |
| 16:27.54 | wasanzy | [TK]D-Fender: in the tcpdump, am seeing some 404 not found error that is why am confused |
| 16:28.08 | [TK]D-Fender | PB it |
| 16:28.27 | [TK]D-Fender | cbeyerlein: bad aim, sorry |
| 16:28.45 | wasanzy | [TK]D-Fender: me? |
| 16:28.50 | [TK]D-Fender | wasanzy: PB IT |
| 16:29.01 | wasanzy | ok |
| 16:29.04 | vassilux | I have to implement call routing from T2 to a modem V92 attached to my FXS module. It works well with the fax modem but I get some trouble with Multitech V92 modem. Any idea ? |
| 16:29.29 | newtonr | cbeyerlein, post your trace on the asterisk-users forum first and see if someone can identify an issue there |
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| 16:30.10 | cbeyerlein | newtonr: k, will try that first |
| 16:31.03 | wasanzy | Can't I attached file to pastbin? |
| 16:31.25 | newtonr | cbeyerlein, asterisk-users mailing list. Still asleep! |
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| 17:27.37 | bibz | exten => _+.,n,GotoIf(${DIALSTATUS}= CHANUNAVAIL | ${DIALSTATUS} = CONGESTION | ${DIALSTATUS}=BUSY]?mw1-warteschlangen,${EXTEN},1) |
| 17:27.44 | bibz | why isnt this line working? |
| 17:28.10 | [TK]D-Fender | because you didn't balance out your expression |
| 17:28.14 | [TK]D-Fender | count your braces |
| 17:28.57 | bibz | oops, one missing |
| 17:30.29 | [TK]D-Fender | Your spacing is also inconsistent across all 3 and have no protections against the variable being blank. Not that those are "current" problems.... |
| 17:34.37 | bibz | exten => _+.,n,GotoIf($[$["${DIALSTATUS}" = "BUSY"] | $["${DIALSTATUS}" = "CONGESTION"]]?mw1-warteschlangen,${EXTEN},1) should do the trick now |
| 17:35.10 | WIMPy | no |
| 17:35.28 | WIMPy | Oh, actually, yes. |
| 17:35.36 | bibz | why not? "BUSY" = "BUSY" not? |
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| 17:35.48 | bibz | oh ok :) |
| 17:35.56 | WIMPy | There are so many brackets to count... |
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| 17:44.14 | [TK]D-Fender | And they're bad |
| 17:44.22 | [TK]D-Fender | no need to nest like that |
| 17:46.40 | drmessano | I told my ex wife the same thing |
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| 17:54.12 | bibz | how is it possible for "NOANSWER" to exist with 0 seconds duration ? |
| 17:54.44 | [TK]D-Fender | bibz: Lets see the call |
| 17:54.56 | WIMPy | Someone cancelled the call after less than a second. |
| 17:58.19 | [TK]D-Fender | That wouldn't respond "noanswer" |
| 17:58.26 | [TK]D-Fender | that' come back "busy", etc |
| 17:58.37 | [TK]D-Fender | "noanswer implies a timeout/fail |
| 18:00.28 | WIMPy | CANCEL rather. So the timeout is the only answer? |
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| 18:32.34 | dan_j | Sorry for the rookie question but i dont usually get involved with ISDN lines. When connecting to a public ISDN line, do I need a BRI or PRI card? |
| 18:33.13 | WIMPy | The one that fits your line. |
| 18:34.05 | WIMPy | BRI=2 B-Channels, PRI=30 B-Channels or 23 or 24. |
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| 18:35.13 | newtonr | dan_j, what country are you in? |
| 18:35.19 | dan_j | UK |
| 18:35.40 | newtonr | Ah okay, yeah you might have either there. |
| 18:36.10 | WIMPy | Did you order isdn2e oder isdn 30? |
| 18:36.18 | dan_j | WIMPy: Ah, I thought PRI and BRI refered to whether you were connecting to a PSTN or to another digital device. |
| 18:36.32 | dan_j | No, they are standard isdn2e lines. |
| 18:36.39 | WIMPy | No, that NT/LT or TE. |
| 18:36.49 | WIMPy | That's a BRI then. |
| 18:37.44 | dan_j | In short, i need a card that can receive 2 ISDN connections (4 lines), and then a 3rd ISDN connection to connect to a fax server. Fax calls are routed through Asterisk to the fax server. |
| 18:38.13 | WIMPy | So a quad BRI card. |
| 18:38.27 | dan_j | WIMPy: So if I spec sheet says 'Supports 2 to 4 ports. Mix TE and NT modes, as required'. That will do? |
| 18:38.39 | WIMPy | Why don't you use your Asterisk box as fax server? |
| 18:38.58 | WIMPy | 2 to 4? |
| 18:39.26 | dan_j | Its an annoying client. I'd prefer to keep the setup as similar to the current setup as possible. |
| 18:40.49 | WIMPy | Then just get a quad BRI card and keep it as is. |
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| 18:44.03 | dan_j | Perfect. Thanks for clearing that up. |
| 18:44.58 | WIMPy | And get a generic card. |
| 18:45.06 | WIMPy | That way you have free choice of drivers. |
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| 19:21.19 | bibz | [TK]D-Fender: http://pastebin.com/qpp01XNg |
| 19:21.29 | bibz | regarding the 0 second "NOANSWER" calls... |
| 19:23.08 | WIMPy | If it wasn't german, I would thing that's FreePBX. |
| 19:23.44 | WIMPy | think |
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| 19:24.52 | bibz | nope, the whole dialplan (2000lines) was being written by some old dude who left the company.. and after 6 months I'm still fixing errors .. |
| 19:25.08 | WIMPy | Ugh |
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| 19:25.23 | [TK]D-Fender | [2015-01-27 20:06:44] VERBOSE[9105][C-0000031a] app_dial.c: -- Called SIP/pbx3-out/+43664xxxxxxxx |
| 19:25.25 | [TK]D-Fender | [2015-01-27 20:06:46] VERBOSE[9105][C-0000031a] pbx.c: == Spawn extension (mw1-rauswaehlen, +43122xxxxxx0305, 7) exited non-zero on 'SIP/pbx3-0000043c' |
| 19:25.26 | [TK]D-Fender | call aborted |
| 19:25.39 | [TK]D-Fender | You'd have to look at SIP debug for a better reason |
| 19:26.37 | WIMPy | Or add some Verbose after that Dial. |
| 19:26.57 | bibz | what do you mean, verbose after dial? |
| 19:27.46 | WIMPy | The dialplan ends after trying to Dial(). You could add an Verbose() to show things like DIALSTATUS, HANGUPCAUSE, etc. |
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| 19:28.05 | bibz | oh okay. thats a good idea actuallz. |
| 19:28.07 | bibz | actually* |
| 19:29.46 | WIMPy | Looks like the h extension found out the last message was "100 Trying". |
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| 19:32.24 | [TK]D-Fender | SIP DEBUG. |
| 19:32.27 | [TK]D-Fender | Accept no substitutes |
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| 19:34.42 | bibz | I'm trying to figure out how to get the SIP debug of that call, since its not in /var/log/asterisk/full... |
| 19:35.25 | WIMPy | Live |
| 19:36.06 | bibz | well.. it was a customer who called, so I can't ask him to retry.. probably it would work by now.. |
| 19:36.51 | WIMPy | Well, if you don't have the information... |
| 19:37.26 | [TK]D-Fender | "probably" and an empty sack is worth the sack.... |
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| 21:44.10 | AnonGirl | Anyone here know what backofficesports.com is? |
| 21:50.17 | WIMPy | Looks like a domain name. |
| 21:51.25 | newtonr | I concur |
| 21:52.05 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:56.11 | AnonGirl | I mean, I think I saw it trying to SIP phreak me |
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| 22:03.04 | Katty | TKFENDER |
| 22:03.15 | Qwell | KATTY |
| 22:03.16 | Katty | ohai. |
| 22:03.18 | Katty | hugs Qwell |
| 22:03.48 | [TK]D-Fender | Katty, Mew. |
| 22:03.58 | Katty | [TK]D-Fender: herro. |
| 22:04.09 | MaliutaLap | wonders who else Katty is trying to rope into the squirrel conspiracy ;) |
| 22:04.23 | Katty | MaliutaLap: who told you to say that?!?! why did you want to join? |
| 22:04.52 | MaliutaLap | Katty: no, I'm the one outing the conspiracy - remember? ;) |
| 22:04.58 | AnonGirl | and now for the fun part |
| 22:05.31 | Katty | MaliutaLap: oh. right. what if i give you a cookie? |
| 22:05.42 | Katty | MaliutaLap: if you're busy eating, you can't talk.... right? |
| 22:06.34 | MaliutaLap | Katty: Oh, I'll take the cookie (it's possible I'm related to Cookie Monster). I can eat and type at the same time :P |
| 22:18.52 | Katty | hehehe |
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| 22:22.08 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:26.08 | AnonGirl | prods malcolmd |
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| 22:27.45 | Visage | SIP ...phreak? Did we just have a collision of past and present realities? I think I just heard the space-time continuum scream. |
| 22:28.20 | AnonGirl | Visage, *giggle* |
| 22:28.27 | malcolmd | eh? |
| 22:29.18 | AnonGirl | malcolmd, you might remember me as the weirdo who at 11 or 12 tried to run an Asterisk server out of their bedroom. |
| 22:29.47 | malcolmd | i don't recall, sorry :) |
| 22:29.50 | AnonGirl | I'm now 14, and now running a YATE server out of some datacentre in France. |
| 22:29.54 | AnonGirl | malcolmd, ok |
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| 22:39.26 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:39.49 | malcolmd | and...that's better :) |
| 22:40.04 | Penguin | Feels exactly the same to me. |
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| 23:52.42 | AnonGirl | malcolmd, and what's better? |
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