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| 01:17.41 | *** join/#asterisk costal (~ivan@corpnat.comindico.com.au) |
| 01:17.46 | costal | hello all |
| 01:18.00 | costal | I'm testing asterisk + realtime mysql for sip users |
| 01:18.21 | costal | I'm wondering if there is a benchmark tool for realtime with mysql |
| 01:18.31 | costal | ? |
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| 02:12.52 | jonny330 | has anyone in here been able to install vicidial on an xen virtual server? |
| 02:13.39 | jonny330 | i don't know if i have a problem with dahdi or not i wanted to talk to someone to see if they can work it out iwth me |
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| 03:01.15 | capitan2 | damn... it fixed itself :( |
| 03:01.52 | WIMPy | Break it agin. Fast! |
| 03:01.56 | capitan2 | all day it's been borked... i finally get in here and it's fixed :( |
| 03:02.09 | WIMPy | What? |
| 03:02.09 | capitan2 | WIMPy: haha i don't know how |
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| 03:02.49 | capitan2 | well i'm not sure... so now i can only ask about whether certain things are "normal" or not... |
| 03:03.33 | WIMPy | Things that fix themselves go into paranormal, I guess. |
| 03:03.36 | capitan2 | here's one thing i was seeing on the console after a few -ddd's... REGISTER attempt 872 to xxxxxx@sip.flowroute.com |
| 03:05.15 | capitan2 | eventually it disconnected me by itself from the console... and now the phones work and i'm no longer seeing that |
| 03:05.53 | capitan2 | WIMPy... paranormal i don't mind... paranormal that waits all day until i'm watching it... that's just spooky ;) |
| 03:07.26 | WIMPy | So you had one kind of network issue. Then another network connection broke and the first one was fixed? |
| 03:08.05 | capitan2 | no... the network connections seem fine |
| 03:08.07 | ChannelZ | Nevermind the things you can't see... Ceiling cat is watching you masturbate. |
| 03:09.04 | capitan2 | what i meant was, the asterisk CLI disconnected for some reason, and everything fixed itself |
| 03:09.21 | capitan2 | ChannelZ: the ceiling cat is why i masturbate in the first place ;) |
| 03:10.13 | capitan2 | wait... that doesn't sound too good... scratch that... |
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| 03:36.10 | ChannelZ | A feline exhibitionist |
| 03:41.07 | leifmadsen | ~seen ctooley |
| 03:41.11 | infobot | ctooley <~ctooley@77.sub-75-222-72.myvzw.com> was last seen on IRC in channel #asterisk-doc, 113d 8h 8m 5s ago, saying: 'Network monitoring is a rather waste of time. :-)'. |
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| 03:48.38 | ChannelZ | Quick, hack him while he's not looking |
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| 04:01.33 | sawgood | Any tips for 'why' this happens: Using a SIP trunk, DTMF works just fine (95% of the time) to non 800 series numbers, but when I call (800) series numbers, DTMF seems to not work 8 out of 10 times |
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| 04:05.23 | ChannelZ | if you're doing rfc to your itsp, then it's their problem/fault |
| 04:07.10 | sawgood | I am using RFC2833 ... (and recently I did switch SIP providers) |
| 04:07.39 | sawgood | What 'ties' this concern to (800) numbers vs non (800) numbers? |
| 04:08.02 | ChannelZ | ?? |
| 04:23.25 | jonny330 | how do i force dahdi to compile dhadi_dummy |
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| 04:25.55 | newmyth | AAAARRRRGGGGGGGGGGGGGGGGG |
| 04:26.23 | newmyth | I just hate it when the only way to fix * is to break * |
| 04:26.29 | WIMPy | jonny330: It's int the core module. |
| 04:26.42 | jonny330 | so it alwasy should be complied? |
| 04:27.03 | WIMPy | yes |
| 04:27.38 | jonny330 | i am comlpiling it but it does not get compiles |
| 04:29.48 | jonny330 | where would the compiled module get loaded? |
| 04:30.36 | p3nguin_ | It was my understanding that 1.8 no longer uses dahdi_dummy. Is that what you're using? |
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| 04:31.05 | jonny330 | asterisk 1.4 |
| 04:32.12 | p3nguin_ | Install dahdi and dahdi-tools. In asterisk source, make clean, then ./configure && make menuselect, followed by choosing the necessary items in the menu, then make && make install. |
| 04:32.54 | jonny330 | i am installing from the complete |
| 04:33.09 | jonny330 | so it has the tools and linux together |
| 04:35.05 | jonny330 | after i do make, i do a search for dahdi_dummy* and all that it finds is dhadi_dummy.c |
| 04:36.26 | p3nguin_ | What kind of search, exactly, are you doing? |
| 04:36.43 | jonny330 | find / -name dahdi_dummy* |
| 04:36.49 | sawgood | Is there a way in which I can 'test' DTMF via SIP (to see SIP messages) for this ... make a call, press a combination of digits, and see if the combination of digits show up in the SIP messages? |
| 04:37.48 | WIMPy | jonny330: That's the way it's supposed to be. Just load dahdi. |
| 04:38.08 | p3nguin_ | Which asterisk version and dahdi version are you using? There was a change in the later versions. |
| 04:39.58 | jonny330 | 1.4.35 |
| 04:40.10 | jonny330 | 2.3.0.1+2.3.0 |
| 04:42.23 | jonny330 | i guess i will keep messing around with it unless it shouldn't be complied by design |
| 04:44.26 | p3nguin_ | I know 2.4.0 doesn't have a separate dummy module anymore, but I don't remember which version it changed in. |
| 04:44.45 | jonny330 | okay well i guess i will have to loook that up |
| 04:44.46 | jonny330 | thanks |
| 04:46.02 | p3nguin_ | My old 2.3.0 package doesn't have a dummy module in it, so it must have changed at 2.3. |
| 04:46.54 | p3nguin_ | My 2.2.1 package has the dummy module. |
| 04:48.14 | p3nguin_ | So if you follow the basic steps above, and then load the regular dahdi module, everything should be fine. |
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| 06:00.49 | justdave | Anyone know how to manually configure PLAR on a Cisco phone via the xml provisioning files? The only docs I can find talk about setting it up in Call Manager, and we don't use that. |
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| 07:07.53 | schmidts | good morning |
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| 07:13.52 | kaldemar | sawgood: did you figure it out already? |
| 07:14.15 | sawgood | Hi ... yes ... Asterisk 1.6.2.14 was the concern |
| 07:14.34 | sawgood | Asterisk 1.6.2.9 works, .11 works ... .12 and .14 fail |
| 07:15.18 | kaldemar | what dtmfmode are you using? |
| 07:15.23 | sawgood | rfc2833 |
| 07:20.48 | kaldemar | then the DTMF is sent in RTP, not as SIP messages. rtp debug will show "Got RTP RFC2833 from..." if you get DTMF from the phone. |
| 07:25.51 | sawgood | thank you! |
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| 07:43.40 | jkroon | hi guys, i've been trying to figure this one out for a while now, the DIALSTATUS variable contains one of a few values, most of these are pretty obvious but I'm having a hard time figuring out CHANUNAVAIL and CONGESTION. What exactly is the difference between these two? |
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| 07:51.47 | schmidts | CONGESTION means impossible and CHANUNAVAIL means not at the moment IMHO |
| 07:52.32 | schmidts | s/not at the moment/possible but not at the moment/ |
| 07:53.13 | WIMPy | Na, CONGESTION can mean round about anything. |
| 07:54.04 | WIMPy | But I don't know when exactely to expect CHANUNAVAIL. |
| 07:55.52 | schmidts | for me it means if you dial for example DIAL(SIP/123) and you get a chanunavail then peer 123 is not reachable / not registered if you get a congestion there is not even a peer 123 |
| 07:57.03 | jkroon | ?!? ok, that's confusing all way round. |
| 07:57.34 | WIMPy | Yes, a non-existing peer is one of the possibilities AFAIR. |
| 07:58.03 | WIMPy | If you want to know more, check HANGUPCAUSE. |
| 07:58.13 | jkroon | basically I've got two upstream providers, and obviously certain codes (eg SIP code 404, or ISDN 1) is terminal, whether I phone over one link or the other that won't change, however, something like ISDN cause code 34 (channel unavailable) might result in success over the alternative link. |
| 07:58.59 | jkroon | Yes, the HANGUPCAUSE codes are very valuable, but there are 128 odd of them and I really don't feel like sifting through all of them and deciding which ones I can retry over alternative links and which ones not. |
| 07:59.34 | WIMPy | I don't think there is an easier way, other than just trying. |
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| 08:00.20 | sawgood | Is there a place at Digium where I could post a note about DTMF not working for me under 1.6.2.14, but that it does work fine under 1.6.2.11 or earlier? |
| 08:00.30 | sawgood | how does Digium keep up with these things you think? |
| 08:00.42 | WIMPy | The location would be valuable in that decision, but that's not available in any variable. |
| 08:01.05 | jkroon | So do a Dial(SIP/prov1/${EXTEN}), and always follow that with a Dial(SIP/prov2/${EXTEN}) if DIALSTATUS is either of CONGESTION or CHANUNAVAIL? that doesn't make sense unless you just want to wait longer for an error message ... |
| 08:01.06 | WIMPy | sawgood: issues.asterisk.org |
| 08:02.44 | WIMPy | Actually you never know if HANGUPCAUSE is set correctely, so maybe that's the best effort anyway. |
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| 08:12.26 | sawgood | Thank you ... I posted what I discovered ... to the issues.asterisk.org page |
| 08:14.35 | jkroon | WIMPy, why would HANGUPCAUSE not be set correctly? |
| 08:15.44 | WIMPy | There are many possibilities if not everything from you to the callee was ISDN. |
| 08:15.46 | kaldemar | sawgood: you should post whether you see the DTMF on the asterisk side. |
| 08:16.04 | WIMPy | Just translate to SIP and back again. |
| 08:16.04 | kaldemar | sawgood: also how you connect to PSTN. |
| 08:16.22 | sawgood | oh ... thanks for the tips (that was my first post) |
| 08:16.52 | sawgood | kaldemar: since the DTMF is in the RTP packets ... is the only way that I could 'see' them is via a Wireshark capture? |
| 08:16.57 | sawgood | Or, can I do something at the console? |
| 08:17.20 | kaldemar | sawgood: you can use rtp debug in CLI. |
| 08:17.34 | sawgood | neat I've not done that ... I'll try that now |
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| 08:18.18 | kaldemar | sawgood: core show help rtp set debug |
| 08:19.22 | sawgood | I ran rtp set debug on |
| 08:19.31 | sawgood | I could see all the RTP stuff fly by ... |
| 08:19.36 | sawgood | I could tell when I pressed a key |
| 08:20.08 | sawgood | Is there any way to view these messages? |
| 08:20.15 | sawgood | I mean they do not give much information |
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| 08:21.56 | sawgood | I could tell when I pressed a key (which key I pressed) ... |
| 08:25.55 | kaldemar | sawgood: they give all kinds of information. event is the key you pressed. |
| 08:26.22 | kaldemar | wireshark will show it in a more friendly format. |
| 08:27.41 | kaldemar | anyhow, if you see asterisk receiving the dtmf and sending it further, there shouldn't be a problem with asterisk. |
| 08:29.35 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
| 08:30.00 | kaldemar | sawgood: how are you connecting to PSTN? |
| 08:30.01 | v1s | is it possible to pass variables to a 2nd ast server ? |
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| 08:30.22 | kaldemar | v1s: over a call, yes. |
| 08:30.40 | ChannelZ | See dtmf in logger.conf |
| 08:31.39 | WIMPy | v1s: core show function iaxvar. Otherwise you have to encode it into the extension. |
| 08:32.30 | kaldemar | or add an X-header in SIP with app SIPAddHeader and get it with func SIP_HEADER. |
| 08:33.08 | v1s | thanks Ill try those methods. |
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| 09:30.10 | SeTTleR | hi, i am trying to connect a siemens gigaset c47h to a tdm400p. how do i connect the tip/ring pins of the tdm400p to the a/b pins of the phone? does anybody know how these two relate? i can build my own cable, i only need to know how :D |
| 09:34.01 | WIMPy | If it's not standard 3/4, it's 2/5. |
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| 09:55.13 | SeTTleR | well the tdm400p manual says, 3=tip and 4=ring and for the siemens phone, i find 2=b, 3=E, 4=W and 5=a |
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| 09:55.53 | SeTTleR | so then would be tip = b, since i should connect 3 with 2 and ring = a?! |
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| 09:56.49 | WIMPy | I can't tell you what is what, but a nd b are tip and ring. |
| 09:57.08 | WIMPy | But unless you need E or W that shouldn't matter. |
| 09:58.45 | SeTTleR | aha? isn't there a direct current between these pins? then it should matter or are the phones protected for that? |
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| 09:59.53 | florz | they just don't care about polarity |
| 10:00.35 | WIMPy | The only devices I know of, where polarity matters, are those automatic precedence switches. |
| 10:01.31 | SeTTleR | aha ok, then i'll give it a try. i always thought that polarity matters there and i couldn't find any information about that. thank you |
| 10:02.31 | florz | the AWADo? =:-) |
| 10:02.44 | WIMPy | jepp |
| 10:04.32 | WIMPy | And if you're one of those guys who use the current on the line to charge your batteries, it will matter as well :-) |
| 10:04.45 | florz | *gg* |
| 10:05.23 | florz | in particular when the power comes from your own PBX ;-) |
| 10:05.26 | SeTTleR | lol who does that? |
| 10:05.50 | SeTTleR | but it's a good idea to charge my batteries with the current of my telco.. would that work? |
| 10:05.56 | WIMPy | Errm, no comment. |
| 10:06.38 | SeTTleR | :D |
| 10:08.35 | florz | consider though that the energy the line would deliver over the course of a year(!) would cost 1 EUR or so when you buy it the usual way ;-) |
| 10:09.27 | SeTTleR | well, 1 eur saved then :D |
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| 10:11.02 | WIMPy | If there wasn't that little catch that the switch will cut the power if the current is above the idle limit for some time without an active call. |
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| 10:18.59 | magwas | which modules are needed for confbridge in 1.6.2? |
| 10:19.10 | v1s | on an outbound context I think I am not understanding it fully i have it start with s,1,verbose(test) then next line I have 6001,1,dial(blah) and next line 6002,1,dial(blah) but it is always skipping the s extension when I dial an extension is there like a catch all or do I have to put verbose to every extension I tried _. but ijust skips the extension and just shows the verbose |
| 10:19.17 | magwas | unloaded all unnecessary modules, and apparently some necessary ones also |
| 10:20.04 | magwas | v1s you sure you don't have another s,1? |
| 10:20.16 | WIMPy | v1s: You only hit s if you didn't dial anything, like on a pots port. |
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| 10:20.41 | v1s | i have include => and in there there is also s,1 |
| 10:21.01 | v1s | do I need to change the includes to s,n or something ? |
| 10:21.58 | AdvoWork | why would one sip extension not register when all others are fine? it just wont register at all, qualify=no, sip show peers shows: (Unspecified) D N 0 Unmonitored.. |
| 10:22.04 | WIMPy | Wow. Even I didn;t have the idea to just include some priorities, instead of full extensions. |
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| 10:23.03 | magwas | in join_conference_bridge: Conference bridge '1234' could not be created. |
| 10:23.36 | v1s | WIMPy: so its bad idea to do it my way I should include the full extensions ? |
| 10:24.20 | WIMPy | v1s: I have no idea what happens if you try. But if you find out, tell us. |
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| 10:29.02 | kaldemar | AdvoWork: is it trying to register? does asterisk see register messages from the phone? check with sip debug in CLI. if not, check the phone's configuration. |
| 10:29.38 | AdvoWork | kaldemar, ive done a log, and sip trace on the phone itself, how would i debug the specific extension? |
| 10:29.50 | AdvoWork | kaldemar, is it worth me pastebining the logs? |
| 10:30.29 | kaldemar | AdvoWork: by the ip address. "sip set debug ip..." |
| 10:30.38 | kaldemar | AdvoWork: if you want some help, you better pastebin. |
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| 10:32.20 | AdvoWork | kaldemar, this is the log: http://pastebin.com/YWn55pvP |
| 10:32.31 | AdvoWork | just upping the trace now |
| 10:33.13 | AdvoWork | heres the trace: http://pastebin.com/p7582Y4j |
| 10:34.03 | v1s | WIMPy: well here is what I guess seemed to work for me. I had to make one context with _. to catch all. Put what I want there. then last line of that I use goto exten and goto another context which calls the included extension. :/ |
| 10:34.49 | v1s | for some reason if I put the include extions in the same context ast seems to go crazy like its in a loop |
| 10:35.10 | kaldemar | AdvoWork: is the ip address correct? does asterisk see the messages? |
| 10:35.33 | kaldemar | v1s: _. is very bad practice, since it matches to all standard extensions. |
| 10:36.04 | kaldemar | v1s: if you want to match all numbers, use _X. |
| 10:36.15 | AdvoWork | kaldemar, the ip is correct, ive got a few phones the same at this other site, theyre all setup the same, theyre all ok. ive tried sip set debug, its asking for ip host:port the host isnt available though according to sip show peers, can i debug based on the extension? |
| 10:37.04 | kaldemar | AdvoWork: use an ip address. the phone must have one. |
| 10:37.26 | v1s | kaldemar: i want it to match everything. in the next context it is matching only the extension. theres not really any dialing going on in that context only setting variables and showing some info. You think its still bad to use there? |
| 10:38.07 | kaldemar | v1s: yes. |
| 10:38.51 | AdvoWork | kaldemar, yea. also ive tried pinging the ip of the router from a pc, it can see that. so if the ip of the phone is 0.59 its sip set debug 192.168.0.59, i try that and it gives: Enables dumping of SIP packets for debugging purposes - sip set debug ip <host[:PORT]> - Enables dumping of SIP packets to and from host..... - sip set debug peer <peername> - Enables dumping of SIP packets to and from host. - Require peer to be registered. |
| 10:38.53 | v1s | kaldemar: so its better to set the same variables a ton of time for each extension |
| 10:39.05 | kaldemar | v1s: you will most likely run into unexpected behavior with _. |
| 10:39.13 | v1s | kaldemar: what is the actual security risk to it though. |
| 10:39.14 | kaldemar | v1s: no, use a proper pattern. |
| 10:39.45 | WIMPy | kaldemar: Not if you always goto another context. |
| 10:40.01 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 10:40.05 | kaldemar | v1s: when you hang up a call, the hangup extension (h) gets called. your extension matches to it and so on. |
| 10:40.32 | v1s | so _X. is better to use |
| 10:41.30 | WIMPy | If you're sure you only want to match extensions starting with a digit, yes. |
| 10:42.29 | v1s | doesnt as soon as it goes to anouther context use the h there ? |
| 10:42.57 | WIMPy | Yes. Hence my comment about the goto. |
| 10:44.05 | *** join/#asterisk ketas- (~ketas@ketas6-sixxs.si.pri.ee) |
| 10:44.24 | v1s | so it maybe bad practice but not necessarily a security risk |
| 10:45.02 | WIMPy | Not if you always goto another context. |
| 10:45.28 | v1s | got it thanks |
| 10:47.04 | WIMPy | So if you have a context with _> as the only extension whe you just do some preparations and the goto another context containing the real extensions, that should be fine. |
| 10:47.52 | AdvoWork | kaldemar, on the phone when i re register, normally it shows up on the cli (asterisk -rvvvv) yet this one, nothinghappens |
| 10:49.49 | kaldemar | AdvoWork: use sip debug by ip address to find out if asterisk sees the messages. |
| 10:52.07 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 10:53.53 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 10:55.46 | AdvoWork | kaldemar, i keep trying that, sip debug ip does nothig |
| 10:56.52 | kaldemar | you need to use the correct command and give it the ip address of the phone. sip set debug ip <ip_address> |
| 11:01.08 | AdvoWork | kaldemar, thats exactly what i tried, it just echos that same information to me |
| 11:01.09 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 11:02.50 | kaldemar | AdvoWork: you pasted something else... |
| 11:03.11 | kaldemar | AdvoWork: sip set debug ip 192.168.0.59 |
| 11:04.21 | *** join/#asterisk Yedidya (~chatzilla@host86-132-230-156.range86-132.btcentralplus.com) |
| 11:08.02 | v1s | sorry if this is a little off topic but has any one used a nom 870? If so are they any good? |
| 11:08.08 | v1s | snom |
| 11:08.57 | WIMPy | No idea, but I like the old ones. |
| 11:18.53 | *** join/#asterisk macno (~macno@2a02:4d0:4:e0a:219:99ff:fe62:597c) |
| 11:20.20 | macno | Hello, I have a problem (1/2 times a week) with "asterisk.c: No more connections allowed" |
| 11:20.39 | tzafrir_laptop | macno, hmm... sounds familiar |
| 11:20.41 | *** join/#asterisk ketas (~ketas@ketas6-sixxs.si.pri.ee) |
| 11:20.47 | macno | In full log I see a Remote Unix Connection every 5 minutes |
| 11:21.10 | macno | magically, at a random point |
| 11:21.16 | tzafrir_laptop | I've seen this somewhere. for starters: ls -l /proc/PID_OF_ASTERISK/fds |
| 11:21.28 | tzafrir_laptop | hmm... wrong command... |
| 11:21.32 | macno | I don't see disconnections, only connections |
| 11:21.39 | tzafrir_laptop | lsof /var/run/asterisk/asterisk.ctl |
| 11:21.54 | macno | untill the AST_MAX_CONNECTIONS is reached (128 ) |
| 11:21.59 | tzafrir_laptop | Do you see many open FDs by the process of asterisk? |
| 11:22.17 | macno | Actually no |
| 11:22.24 | macno | but thanks! That's a good hint |
| 11:22.29 | tzafrir_laptop | Looks like some sort of leak. asterisk does not close that file descriptor |
| 11:22.42 | macno | yeah something like that |
| 11:22.44 | tzafrir_laptop | I ran into this somewhere, but failed to reproduce it |
| 11:23.00 | macno | but what is that connect every 5 minutes? |
| 11:23.17 | tzafrir_laptop | flash operator panel or some other monitor? |
| 11:25.34 | macno | I should see it when I have the problem using lsof |
| 11:26.12 | tzafrir_laptop | If you see some 128 different open file descriptors from the asterisk process for that file, yes |
| 11:27.43 | macno | I opened a asterisk -r from another console |
| 11:27.51 | macno | and now I see 2 fd |
| 11:29.28 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:30.37 | macno | but I can't see who opened it. process is always asterisk. |
| 11:42.22 | *** join/#asterisk stix (~stix@firewall.o4.dk) |
| 11:43.25 | stix | Hi guys. The command queue show <queue> gives me some stats. But I cannot find it in the documentation. Can anyone tell me what: W:0, C:7, A:3, SL:0.0% is? |
| 11:44.46 | *** join/#asterisk Yedidya (~chatzilla@host86-132-230-156.range86-132.btcentralplus.com) |
| 11:45.02 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 11:55.49 | *** join/#asterisk fofware (~Fabian@host186.190-225-12.telecom.net.ar) |
| 12:07.16 | *** join/#asterisk Crashvr (~arthur@200-170-196-78.core01.spo.ifx.net.br) |
| 12:13.46 | Crashvr | Could anyone inform me whether there is a way to save the log's sip debug on files? |
| 12:15.40 | ectospasm | Crashvr: try running the CLI through tee: asterisk -vvvrnT | tee /tmp/astlog |
| 12:16.43 | *** join/#asterisk fofware (~Fabian@host186.190-225-12.telecom.net.ar) |
| 12:17.00 | jbroome | didn't this come up yesterday AM too? |
| 12:17.33 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 12:20.22 | *** join/#asterisk WindBack (~quassel@kirk.capitalinasdc.com) |
| 12:21.40 | *** join/#asterisk boch (~fseratti@190.220.65.19) |
| 12:23.04 | *** join/#asterisk jmkgreen (~chatzilla@wish-hq3.gotadsl.co.uk) |
| 12:23.23 | jmkgreen | here's a tricky problem for someone awake enough to solve |
| 12:23.44 | jmkgreen | we call from Asterisk to a customer - we don't get the DTMF tones he inputs |
| 12:23.55 | jmkgreen | if he calls our Asterisk, we do hear his DTMF tones |
| 12:24.19 | jmkgreen | we're baffled :-) |
| 12:24.24 | *** join/#asterisk dimm (~miniadmin@unaffiliated/dimm) |
| 12:25.23 | dimm | hello. i get new job. now i have treouble with asterisk. can't dial some number |
| 12:25.35 | dimm | what be a good start point for diagnostics ? |
| 12:26.12 | boch | see consoles output |
| 12:27.05 | leifmadsen | jmkgreen: SIP being used in both directions? |
| 12:28.08 | jmkgreen | leifmadsen: believe so. We're using sip to talk to our VoIP provider. That's really as far as we go on that level. |
| 12:28.19 | jmkgreen | we're dialling him through that provider |
| 12:28.50 | jmkgreen | leifmadsen: customer is on a UK landline in a company building somewhere. No further information available to us. |
| 12:29.46 | jmkgreen | leifmadsen: we have had similar reports that our system has ignored people's DTMF tones but they represent a small proportion of our messages sent and are proving impossible to debug on their own. |
| 12:30.25 | v1s | jmkgreen: what dtmfmode are u using? |
| 12:30.30 | v1s | did u try some of the others? |
| 12:30.41 | jmkgreen | vls: we have tried rfc2833 and inband. |
| 12:31.08 | jmkgreen | we also record our test calls to this particular customer and we hear no DTMF tones but he says he can hear them on his own handset |
| 12:31.44 | jmkgreen | in the last call we used alaw, if that's of any help |
| 12:32.03 | jmkgreen | we also know it happens through two different VoIP providers |
| 12:33.07 | jmkgreen | and we have Asterisk 1.4 and 1.6 installations, both of which have had the issue. It's almost as though every customer who's affected are using some broken PBX equipment, but we are generally targetting people's home landlines making that a seemingly unlikely scenario. |
| 12:34.00 | leifmadsen | jmkgreen: you might also wish to try relaxdtmf=yes |
| 12:35.07 | leifmadsen | jmkgreen: also, when it does happen, you might wish to capture the calls with wireshark and analyze what the network is seeing to see if it's an asterisk problem, or if there really is no DTMF (or perhaps DTMF that isn't following protocol, etc.) |
| 12:35.18 | v1s | can also try  toneduration=300 |
| 12:35.49 | leifmadsen | ok, I'm off to write some dialplan for the new book. I'll be back later. |
| 12:35.56 | jmkgreen | leifmadsen: we're fairly confident it isn't asterisk as an identical call to our office landline works fine |
| 12:36.24 | leifmadsen | jmkgreen: good to know :) I'd look at the wireshark avenue then to verify what you're seeing so you can lay blame on someone else :) |
| 12:37.39 | v1s | any one have an idea what would be the best way to put 3 different trunks in a random order and dial them ? |
| 12:37.52 | v1s | in that order? |
| 12:38.35 | v1s | 3 being more then less then or equal to ;) |
| 12:38.45 | *** join/#asterisk fofware (~Fabian@host186.190-225-12.telecom.net.ar) |
| 12:39.12 | *** join/#asterisk bmg505 (~leon@196-209-120-122.dynamic.isadsl.co.za) |
| 12:39.34 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 12:39.55 | wdoekes2 | isn't a random starting point sufficient? |
| 12:40.55 | v1s | not really because alot of times the call doesnt go through till 8-10times later. |
| 12:41.17 | v1s | and if I put it through to a single trunk over it takes longer |
| 12:42.06 | dimm | boch, console output is on tty where i can see ' * CLI > ' ? |
| 12:42.29 | boch | dimm, yes |
| 12:44.04 | Yedidya | anyone seen issue with incompatible asterisk16-addons-mysql x86_64 1.6.2.1 <=> asterisk16 x86_64 1.6.2.14 |
| 12:44.22 | Yedidya | http://pastebin.com/ATTH2FKk |
| 12:44.42 | Yedidya | log off issue |
| 12:45.25 | v1s | Yedidya: think u using 2 different versions thats why |
| 12:45.34 | v1s | 2.1 and 2.14 |
| 12:46.08 | Yedidya | these were installed form packages.asterisk.org and are the latest. |
| 12:46.22 | Yedidya | via yum |
| 12:50.07 | dimm | is 'asterisk -vvvvvvvvvvvr -g' it is a maximum verbosity level for diagnostics? |
| 12:50.28 | ectospasm | dimm: no |
| 12:50.43 | WindBack | Hello, I have an abnormal problem with some random calls in my Asterisk PBX. Sometimes when I have a call established to the PSTN, it goes down. |
| 12:50.43 | WindBack | The internal extension is SIP while the connection to the PSTN is trough DAHDI using ISDN/PRI. |
| 12:50.43 | WindBack | Debuging SIP and PRI for some of this calls, I could see the following: http://www.pastebin.ca/1994467. |
| 12:50.43 | WindBack | It seems that the PBX is hanging both channels because it is sending a DISCONNECT message to the PSTN and also it is sending a BYE message to the extension. |
| 12:50.43 | WindBack | Can anybody know why the PBX is hanging the call by itself? |
| 12:51.08 | ectospasm | dimm: you'd need sizeof(int) 'v' options to reach that |
| 12:51.18 | ectospasm | dimm: which is a bit absurd |
| 12:51.25 | Yedidya | actualy the latest vers for source is 1.6.2.14, 1.6.2.2 |
| 12:51.44 | ectospasm | dimm: though in practice verbosity doesn't increase past ten or so |
| 12:52.08 | Yedidya | vls: do you know who is responsible for making the rpm packages on asterisk repo? |
| 12:52.25 | dimm | what is it? |
| 12:52.27 | dimm | == Manager 'pbx' logged off from 127.0.0.1 |
| 12:52.28 | dimm | <PROTECTED> |
| 12:52.28 | dimm | <PROTECTED> |
| 12:52.40 | AdvoWork | if ive done: sip set debug ip 192.168.0.59 how do i stop it? |
| 12:52.43 | dimm | it is printing and printing to console |
| 12:53.06 | kaldemar | AdvoWork: sip set debug off |
| 12:53.06 | Yedidya | AdvoWork: sip set debug off |
| 12:53.51 | kaldemar | dimm: some third party application that is connecting to manager interface. |
| 12:54.26 | AdvoWork | kaldemar, i did that for the IP, tried to re register and nothing, it showed nothing at all. kaldemar would this matter that the server is at locationA, and the phone is at location B? (for the debug by that ip I mean?) |
| 12:55.20 | kaldemar | AdvoWork: was that the ip of the phone? |
| 12:55.21 | *** join/#asterisk PoTe (~PoTe@rev-200-40-119-222.netgate.com.uy) |
| 12:55.28 | dimm | kaldemar, it is look as ' logged on.....1sec... logged off ....1 sec logged on .... 1sec ... logged off ' |
| 12:55.32 | dimm | sorry for my english |
| 12:57.29 | AdvoWork | kaldemar, yeah, but im thinking(thats the ip of the phone on the network of location B) but that ip may be different here(at location A) ? |
| 12:58.02 | Yedidya | dimm: that is the FreePBX System Status page. |
| 12:58.37 | kaldemar | AdvoWork: if it's behind a nat, you need to use the address asterisk sees its messages coming from. |
| 12:59.06 | kaldemar | AdvoWork: or just set the debug on for all traffic and look for the messages from your phone. |
| 13:01.00 | AdvoWork | sorry to be dumb, but how can i see what address its coming from, and know its that phone? additionally how would i set debug on for everything? |
| 13:01.10 | *** join/#asterisk luckman212 (~quassel@pool-96-246-172-198.nwrknj.fios.verizon.net) |
| 13:02.28 | kaldemar | ~sipnat |
| 13:02.28 | infobot | rumour has it, sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:04.07 | kaldemar | AdvoWork: you need to know environment you're working in, check the router or do whatever.. "sip set debug on" will show all messages. and see the above guide for configuring your asterisk right. |
| 13:04.55 | dimm | what file is config for rules of dialing numbers? |
| 13:05.05 | kaldemar | dimm: extensions.conf |
| 13:05.09 | dimm | something like "8499xxxxxxx" |
| 13:05.32 | kaldemar | if you're using freepbx, you better not touch it. |
| 13:05.42 | dimm | yes, i'm using freepbx |
| 13:06.11 | dimm | some settings is go out, and now we can dial 8-499, but cannot dial to 8-495 numbers |
| 13:06.35 | kaldemar | ask in #freepbx how you should configure your system. |
| 13:07.51 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 13:08.40 | dimm | ok |
| 13:08.55 | dimm | can firewall be a key to some trouble with asterisk ? |
| 13:09.10 | dimm | we can not dial to another city |
| 13:09.35 | kaldemar | depending on your setup, yes. |
| 13:09.41 | dimm | calling to our city is going via city telephone network, instead internet |
| 13:13.34 | defswork | doubt that that is your firewall |
| 13:13.44 | defswork | that is your dial plan |
| 13:13.54 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:14.42 | defswork | dimm, you have line card and sip ? |
| 13:14.58 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e4.bcn.adamo.es) |
| 13:17.45 | WindBack | Hello, I have an abnormal problem with some random calls in my Asterisk PBX. Sometimes when I have a call established to the PSTN, it goes down. The internal extension is SIP while the connection to the PSTN is trough DAHDI using ISDN/PRI. Debuging SIP and PRI for some of this calls, I could see the following: http://www.pastebin.ca/1994467. It seems that the PBX is hanging both channels because it is sending a DISCONNECT message to the PSTN and |
| 13:17.45 | WindBack | <PROTECTED> |
| 13:17.55 | dimm | defswork, i have sip, and analog line |
| 13:18.27 | defswork | so your dial plan is deciding to route the call down the analog line card |
| 13:19.40 | *** join/#asterisk sque (~sque@77.49.168.115.dsl.dyn.forthnet.gr) |
| 13:20.19 | sque | Hi I am facing a wierd problem. When I make /etc/asterisk symlink to another folder with EXACTLY the same files asterisk loads with ZERO configuration ( no modules, nothing) |
| 13:20.46 | sque | If I create a real folder /etc/asterisk and move the same files in this folder, asterisk loads just fine |
| 13:21.13 | tuxx- | sounds like your symlinks are fscked |
| 13:21.14 | tuxx- | ;p |
| 13:21.22 | tuxx- | working with symlinks here, no problems at all |
| 13:21.57 | dimm | defswork, dial plan setted via freepbx? |
| 13:22.07 | sque | fscked? |
| 13:22.10 | sque | tuxx-, |
| 13:22.12 | tuxx- | screwed |
| 13:22.12 | tuxx- | ;P |
| 13:22.23 | sque | no something else happens |
| 13:22.26 | tuxx- | not correct, wrong etc. etc. |
| 13:22.41 | sque | maybe its debian thing |
| 13:23.06 | sque | anyway I just wanted to know that there is not common and should work, right? |
| 13:23.13 | sque | s/there/this |
| 13:23.51 | defswork | sque, deffo sounds like your symlink was wrong |
| 13:24.13 | defswork | [[ ! -r /etc/asterisk ]] && ln -s /other/directory /etc/asterisk |
| 13:24.30 | sque | Hmm, If I don't use /etc/init.d/asterisk |
| 13:24.42 | sque | and start asterisk manual , with asterisk -d |
| 13:24.44 | defswork | (if /etc/asterisk already existed you would get /etc/asterisk/asterisk ) |
| 13:24.45 | sque | It works |
| 13:24.54 | sque | no files are ok |
| 13:25.09 | defswork | sque, so check the init script - probably checks the dir with -d |
| 13:25.17 | defswork | which (iirc) will fail on a symlink |
| 13:25.17 | sque | sque@sip2:~$ ls -la /etc/asterisk |
| 13:25.18 | sque | total 808 |
| 13:25.18 | sque | drwxr-xr-x 3 asterisk asterisk 12288 Nov 17 15:11 . |
| 13:25.18 | sque | drwxr-xr-x 90 root root 4096 Nov 7 01:56 .. |
| 13:25.18 | sque | -rw-r----- 1 asterisk asterisk 140 Nov 1 00:34 adsi.conf |
| 13:25.20 | sque | 4 lines |
| 13:25.37 | sque | and it has more... files, just wanted to show you that location is ok |
| 13:26.09 | defswork | if ! [ -d /etc/asterisk ] ; then |
| 13:26.10 | defswork | <PROTECTED> |
| 13:26.10 | defswork | <PROTECTED> |
| 13:26.10 | defswork | fi |
| 13:26.18 | defswork | thats in the init script |
| 13:26.40 | defswork | comment it out or correct it to cope with a symlink |
| 13:34.26 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 13:35.09 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:35.18 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:37.14 | *** join/#asterisk [netman] (~netman@34.Red-83-54-35.dynamicIP.rima-tde.net) |
| 13:38.59 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 13:45.19 | *** join/#asterisk nunners (~chatzilla@host81-149-123-240.in-addr.btopenworld.com) |
| 13:45.52 | nunners | Can someone talk a look at http://pastebin.com/bvtXJmjB and give me some clue as to why it doesn't connect to the SIP Extension 211, and drops the call? |
| 13:49.52 | *** join/#asterisk [netman] (~netman@83.54.227.82) |
| 13:50.18 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:55.49 | tzafrir_laptop | defswork, bashism note: use '[' instead of '[[' |
| 13:56.38 | dimm | can i grep from 'asterisk -vvv -r' ? |
| 13:56.53 | dimm | i need only log from one abonent |
| 13:57.22 | tzafrir_laptop | dimm, maybe. But it is generally preffered to grep from the logs |
| 13:57.33 | tzafrir_laptop | tail -f /var/log/asterisk/somelog |
| 13:58.33 | dimm | tail -n 0 -f <path> ) |
| 13:58.46 | WindBack | tzafrir_laptop: tzafrir, can you help me?: I have an abnormal problem with some random calls in my Asterisk PBX. Sometimes when I have a call established to the PSTN, it goes down. The internal extension is SIP while the connection to the PSTN is trough DAHDI using ISDN/PRI. Debuging SIP and PRI for some of this calls, I could see the following: http://www.pastebin.ca/1994467. It seems that the PBX is hanging both channels because it is sending |
| 13:58.46 | WindBack | a DISCONNECT message to the PSTN and also it is sending a BYE message to the extension. Can anybody know why the PBX is hanging the call by itself? |
| 13:58.56 | tzafrir_laptop | defswork, also: ln -s nosuchfile symlink; test -r symlink && ln -s otherfile symlink |
| 13:59.22 | tzafrir_laptop | err... |
| 13:59.33 | tzafrir_laptop | ln -s nosuchfile symlink; test ! -r symlink && ln -s otherfile symlink |
| 13:59.42 | tzafrir_laptop | That would actually try to symlink and fail |
| 14:00.26 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 14:02.25 | tzafrir_laptop | WindBack, your side decided to disconnect in that case, right? |
| 14:03.09 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 14:03.25 | defswork | tzafrir_laptop, you got any eggs ? |
| 14:04.31 | *** join/#asterisk jhirley (~chatzilla@mail.mmdlaw.com) |
| 14:05.56 | WindBack | tzafrir_laptop: I see the PBX desconnecting the call, but the extension neither the target in PSTN hangs the call. So I'm asking why the PBX hanged both channels |
| 14:06.51 | sque | sque, The problem wth symlinks, ti was permissions problem. After moving to a new folder the files changed owner to "root" and it worked only manually because manually I started service as root user, while the init.d scripts start service under "asterisk" user. |
| 14:12.23 | *** join/#asterisk [netman] (~netman@81.Red-83-55-246.dynamicIP.rima-tde.net) |
| 14:14.19 | dimm | situation: two abonents, placed in 8-495-xxxxxxx, but one can be dialed from asterisk, and another - not, can you get some info for this situation? |
| 14:24.25 | *** join/#asterisk shapr (~shapr@nat/digium/x-otjyhpvidijfzils) |
| 14:29.15 | kaldemar | dimm: watch the CLI when a call fails |
| 14:32.59 | *** join/#asterisk [netman] (~netman@246.Red-83-54-218.dynamicIP.rima-tde.net) |
| 14:35.02 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:35.20 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:38.48 | *** join/#asterisk theron (~theron@ip244.scolloc.lh.net) |
| 14:41.08 | *** join/#asterisk carterv (~2panther@63-7.logisoft.com) |
| 14:41.49 | dimm | kaldemar, http://pastebin.com/5zQdg5Yf |
| 14:42.20 | dimm | kaldemar, on line 84 i hear BUSY signal in my headphones |
| 14:42.40 | carterv | is it possible for someone to lend me a hand? I've got a test asterisk box running on CentOS 5.5 and I have xlite for linux on the same box registered but I can't register xlite for windows to the same asterisk box. |
| 14:43.16 | dimm | kaldemar, 117 it is my number in our office telephone network |
| 14:43.35 | carterv | and I'm about 2 days old in the world of asterisk |
| 14:43.47 | atan2 | carterv, windows firewall? |
| 14:44.44 | carterv | I see that it's an allowed program in the firewall exceptions list (windows 7) |
| 14:44.58 | atan2 | Disable the firewall for a moment & try again? |
| 14:45.21 | atan2 | My Windows firewall causes all kinds of troubles. Bleh. |
| 14:45.29 | atan2 | That and nat. Good old nat.... |
| 14:45.50 | carterv | let me try |
| 14:45.56 | atan2 | I suppose get onto your asterisk box and asterisk -rvvvvvvvvvv & see if it even tries to register |
| 14:46.23 | *** join/#asterisk [netman] (~netman@17.Red-81-47-151.staticIP.rima-tde.net) |
| 14:47.15 | carterv | xlite is "enabling account" and I get nothing returned from the CLI |
| 14:47.37 | atan2 | Suffice to say xlite isn't connecting properly |
| 14:47.55 | atan2 | Assuming your settings are correct you could try another client, like Ekiga |
| 14:48.07 | kaldemar | dimm: nothing helpful there, expect the dialed number on lines 82 and 83. is it valid? was it a BRI you have? |
| 14:48.13 | atan2 | In any case I need to fun to make it to class |
| 14:48.17 | atan2 | Peace guys =) |
| 14:49.24 | dimm | kaldemar, is it BRI it is a Basic Rate Interface? |
| 14:49.27 | carterv | thanks atan, I'll try ekiga |
| 14:49.35 | kaldemar | dimm: yes |
| 14:50.46 | dimm | kaldemar, for answer about BRI please wait, i need time for answer |
| 14:51.16 | dimm | kaldemar, i imagine that number is right becouse another number in 660xxxx calling good |
| 14:51.53 | dimm | kaldemar, 6601180 calling good from cellular phone ) |
| 14:52.27 | kaldemar | dimm: your asterisk is not dialing 6601180, but 96601180. |
| 14:52.44 | dimm | kaldemar, yes, it add 9 to number |
| 14:53.08 | dimm | kaldemar, i not use isdn, i use sip |
| 14:53.30 | kaldemar | dimm: DAHDI/g0/96601180 is not sip. |
| 14:53.44 | *** join/#asterisk jkroon (~jkroon@dsl-241-227-21.telkomadsl.co.za) |
| 14:53.53 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-zwgenqsebtcwihhe) |
| 14:54.12 | kaldemar | dimm: either analog, PRI or BRI. is it analog or not? |
| 14:54.52 | *** join/#asterisk phlog1 (~tobias@001de04e4c49.dfn.mwn.de) |
| 14:55.18 | thehar | yawns |
| 14:55.24 | stix | Hi guys. If I use the monitor action on the AMI and set the mix: true, asterisk puts the two wav-files into one, but doesn't mix them. They are played one after another. Am I doing something wrong? |
| 14:56.03 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 15:00.10 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:00.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:01.13 | dimm | kaldemar, 6601180 it is a number in city, another word, it is +7-495-660-1180 |
| 15:02.26 | kaldemar | dimm: 96601180 is what you are seding out. is it valid? |
| 15:02.37 | kaldemar | s/seding/sending/ |
| 15:03.00 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:03.07 | kaldemar | dimm: what about that DAHDI line of yours? is it analog or not? |
| 15:03.35 | dimm | when i call to 96601180 from analog simple phone, i cannot dialing to this number |
| 15:04.12 | dimm | but when i sending out the 96603704 then i calling to +7-495-660-37-04 |
| 15:04.32 | dimm | kaldemar, i not know about DAHDI , sorry |
| 15:05.49 | kaldemar | dimm: you're not making sense now. 96603704 does not equal to +7-495-660-37-04. |
| 15:12.06 | kaldemar | dimm: try command "pri set debug on span 1" in CLI. if you have a digital line, you'll see the protocol output and a cause for the hangup. |
| 15:17.24 | *** join/#asterisk newmyth (~newmyth@68-117-102-144.dhcp.eucl.wi.charter.com) |
| 15:21.16 | stix | Any idea about the monitor action? |
| 15:21.30 | *** join/#asterisk moy_ (~moy@190.108.64.66) |
| 15:22.03 | dimm | kaldemar, our office behind big corporate phone box and 9 it is a output to city line, do you understand this point? ) |
| 15:22.42 | dimm | securityrussia*CLI> pri set debug on span 1 |
| 15:22.42 | dimm | No such command 'pri set debug on span 1' (type 'core show help pri set' for other possible commands) |
| 15:22.42 | dimm | securityrussia*CLI> |
| 15:23.21 | kaldemar | dimm: sure i do, you should have mentioned it half an hour ago. |
| 15:23.21 | dimm | kaldemar, another word, 96601180 it is a right for our case |
| 15:23.22 | p3nguin_ | dimm: Type in "pri set debug" and then press your TAB key to see possible choices. |
| 15:23.53 | kaldemar | dimm: what does a successful call look like? |
| 15:23.59 | dimm | kaldemar, sorry, i know about 9 only now |
| 15:25.22 | kaldemar | looks like your issue is not about asterisk if 96603704 goes through the other system but 96601180 does not. |
| 15:26.02 | dimm | i type 'pri' then press Tab, and 'intense' is auto-typed, if i type 'pri s' <tab> then nothing was add |
| 15:26.35 | kaldemar | pri intense debug span 1 |
| 15:26.46 | p3nguin_ | pri intense TAB? |
| 15:26.48 | dimm | kaldemar, i not understand your question about successful call? must i post log? |
| 15:27.04 | dimm | yes, pri intense debug span |
| 15:27.12 | p3nguin_ | Tab key helps to see what sub commands are available. |
| 15:27.29 | kaldemar | dimm: go ahead, pastebin a successful call. |
| 15:27.30 | dimm | securityrussia*CLI> pri intense debug span 1 |
| 15:27.30 | dimm | No such command 'pri set debug 2 span 1' (type 'core show help pri set' for other possible commands) |
| 15:27.30 | dimm | securityrussia*CLI> |
| 15:27.47 | jkroon | has anybody here ever worked with the SNOM-PA1 or similar? |
| 15:27.59 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 15:28.08 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 15:32.41 | timholum | does anyone know if there is a variable to tell which phone picked up? I have a line exten => Dial( SIP/phone1&SIP/phone2|30|M(macro)) I would like to have the macro do different stuff depending on if phone1 or phone2 picks up, is there any way to do that? |
| 15:33.05 | timholum | ${EXTEN} just tells the dialed exten |
| 15:33.35 | p3nguin_ | You'll probably have to use the channel information. |
| 15:35.06 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:37.09 | timholum | ok I think I found the variable I need DIALEDPEERNUMBER |
| 15:37.33 | timholum | p3nguin_ thanks for the tip, I totaly forgot to look there |
| 15:39.02 | stix | Any of you 200+ users who has ever tried the monitor action on the AMI? |
| 15:41.40 | skrusty | is it possible to log sip debug messages to a specific file? or even to the manager interface? |
| 15:42.56 | p3nguin_ | skrusty: asterisk -rx "sip set debug on" > sip_debug.log |
| 15:43.15 | p3nguin_ | Wait, that's not going to work. |
| 15:43.33 | dimm | kaldemar, http://pastebin.com/ARb71Cf4 it is a successful call (sorry for waiting - another job with users :) ) |
| 15:44.24 | p3nguin_ | skrusty: Did you enable the "full" log and see what debug goes in there? |
| 15:45.32 | skrusty | no, but that would include console messages too, i just want the sip messages; but i guess if that's the only way to do it. |
| 15:47.13 | *** join/#asterisk RAFAIP (~chatzilla@190.145.253.51) |
| 15:47.24 | phlog1 | has anybody any experience in configuring SRTP into asterisk 1.8 or knows some documentation.... |
| 15:47.38 | RAFAIP | hi everyone ^^ |
| 15:47.42 | jkroon | skrusty, no, sip debug logs to verbose, which IMHO is wrong, it should log to debug. You should still be able to get verbose from chan_sip.c log to a separate file, look in logger.conf for ideas. |
| 15:48.00 | skrusty | ok, cheers |
| 15:48.30 | kaldemar | dimm: you don't seem to have any issue with asterisk. |
| 15:48.30 | skrusty | is it possible (via code changes) to log sip debug messages out via a manager event, in theory? |
| 15:48.42 | jkroon | yes. |
| 15:48.52 | skrusty | just looking at better ways of running sip traces |
| 15:49.20 | jkroon | i just log verbose to a file called /var/log/asterisk/verbose and then I've got a script that can parse that to get the SIP conversations out of it. |
| 15:49.36 | skrusty | hmm |
| 15:49.41 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:50.50 | dimm | kaldemar, thx a lot ) |
| 15:50.52 | RAFAIP | i'm trying to access the AMI using PHP, I can log in and log out but the actions GetVar and SetVar doesn't work |
| 15:51.18 | dimm | kaldemar, it can be another trouble with our telephone provider |
| 15:51.51 | RAFAIP | the response that shows me is: ---> Response: Error Message: No variable specified |
| 15:54.35 | jkroon | RAFAIP, command show ???? will show you what needs to be set |
| 15:54.45 | *** join/#asterisk michael-i (~michael-i@141.41.40.223) |
| 15:55.39 | michael-i | Hi everyone. I'm trying to capture missed calls due to the calling party hanging up before the callee can answer. My logic is in a macro using 1.6.2 and the h extension is not being called after hangup. |
| 15:55.52 | michael-i | Which Dial() option am I missing here? |
| 15:56.45 | michael-i | I thought this would be a NOANSWER but that only is activated in my dialplan when the DIAL times out. I need something for when the calling party simply hangs up the phone |
| 16:00.39 | WindBack | tzafrir_laptop: tzafrir, can you help me?: I have an abnormal problem with some random calls in my Asterisk PBX. Sometimes when I have a call established to the PSTN, it goes down. The internal extension is SIP while the connection to the PSTN is trough DAHDI using ISDN/PRI. Debuging SIP and PRI for some of this calls, I could see the following: http://www.pastebin.ca/1994467. It seems that the PBX is hanging both channels because it is sending |
| 16:00.39 | WindBack | a DISCONNECT message to the PSTN and also it is sending a BYE message to the extension. Can anybody know why the PBX is hanging the call by itself? |
| 16:00.58 | WindBack | tzafrir_laptop: sorry tzafrir |
| 16:01.11 | WindBack | tzafrir_laptop: i repeated the same |
| 16:03.01 | timholum | hmm, unfortuantly both DIALEDPEERNUMBER and BRIDGEPEER do not work in my macro, they return blank varialbes? does anyone know how to get which extention picked up and pass it to a variable in Dial(phone1&phone2|30|M(mymacro^${SOMEVARIABLE})) Does anyone know what I would have to change ${SOMEVARIABLE} with? |
| 16:03.46 | timholum | I did a dumpchan at the end of the call and DIALEDPEERNUMBER had what I want but when I use it it comes up blank :( |
| 16:06.09 | RAFAIP | jkroon , i've read it and i'm making it right (I think...) and this is my code: fwrite($socket, "Action: SetVar\r\n\r\n"); fwrite($socket, "Variable: AtestVariable\r\n\r\n"); fwrite($socket, "Value: 999\r\n\r\n"); but still showing error |
| 16:06.38 | russellb | RAFAIP: how many people did you send direct emails to with your AMI question? |
| 16:06.44 | russellb | and you're here, too? |
| 16:07.46 | RAFAIP | xD |
| 16:07.58 | RAFAIP | I need to solve it as soon as I can |
| 16:08.01 | russellb | i'm not smiling |
| 16:08.09 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:08.15 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 16:08.38 | *** join/#asterisk Tim_Toady (~moi@77.49.252.191.dsl.dyn.forthnet.gr) |
| 16:09.07 | jbroome | wait, not PMs, but real emails? |
| 16:09.10 | RAFAIP | there is something wrong? |
| 16:09.11 | russellb | yes |
| 16:09.19 | russellb | i answered your email by the way ... |
| 16:09.21 | jbroome | thats obnoxious as hell |
| 16:09.28 | russellb | jbroome: quite |
| 16:09.41 | russellb | in our internal digium chat room a bunch of people realized that they all just got the same email |
| 16:10.02 | RAFAIP | umm... i'm sorry |
| 16:10.15 | RAFAIP | i didn't realize |
| 16:10.15 | jbroome | my reply to that email would involve a photo of three elderly gentlemen enjoying each others company. nude |
| 16:10.16 | russellb | thanks for distracting everyone with the same question that you should have used the community support forums for (here and the -users list) just like everyone else |
| 16:10.59 | RAFAIP | i've used the forum for search information, but when I post, anyone aswers |
| 16:11.00 | russellb | how did you not realize that sending a direct question to every email address you could find might be distracting and perhaps not the right way to get support for an open source project? |
| 16:11.27 | RAFAIP | i'm sorry! |
| 16:11.29 | russellb | anyway, enough ranting ... i responded to your email, so you should be good. Just don't do that again. |
| 16:11.57 | RAFAIP | all right, thanks a lot anyway |
| 16:12.12 | jbroome | Yeah, my response would be a bit more civil if i was @digium.com. |
| 16:12.16 | RAFAIP | and sorry again, i didn't wantes to be disrutbing |
| 16:18.04 | fullstop | is wondering what goes on in the internal digium chat room. |
| 16:18.28 | fullstop | Parties, puppy dogs and cake, I am sure. |
| 16:18.29 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 16:18.53 | russellb | "anyone want to go out to lunch?" |
| 16:18.55 | russellb | that kind of stuff |
| 16:19.06 | fullstop | russellb: suuuure. ;-) |
| 16:19.18 | russellb | and plotting our next steps for taking over the world? |
| 16:19.25 | fullstop | "anyone want to go out to lunch... on my new rocket ship?" |
| 16:19.27 | russellb | mostly lunch, though |
| 16:19.41 | anonymouz666 | strange situation with dahdi 2.3.0.1 AND 2.4.0 and wct4xxp. |
| 16:20.01 | anonymouz666 | all the E1s went to yellow state and then only after dahdi restart everything goes OK again. |
| 16:20.48 | russellb | hm, weird. you could try digium support ... http://www.digium.com/en/supportcenter/ |
| 16:21.02 | anonymouz666 | it is. |
| 16:21.49 | stix | Where is the official online documentation for asterisk16? |
| 16:21.50 | anonymouz666 | there is a spanish guy on digium support, right ? |
| 16:21.59 | russellb | anonymouz666: yes |
| 16:22.00 | fullstop | better contact them before all of digium heads out in russelb's new rocket ship. |
| 16:22.24 | anonymouz666 | russellb: this is good ! |
| 16:27.17 | AdvoWork | if a phone says NR, can i remotely register that phone if i know the sip extension, or can it only be done from the web interface of that phone? |
| 16:27.39 | *** join/#asterisk trelane (trelane@funtoo/staff/trelane) |
| 16:28.02 | fullstop | AdvoWork: That probably depends on the phone, no? |
| 16:28.09 | michael-i | I just tried adding the 'F' and 'g' flags to my dial command but still nothing. How can I make sure that when callers hangup before I answer, I can capture that for a missed call notification? Something silly I know so sorry for the noise. |
| 16:29.25 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 16:30.33 | *** join/#asterisk myster (~myster@207.229.57.210) |
| 16:33.25 | robl^laptop | anyone have any feedback on Vestec ASR vs Lumenvox ASR under Asterisk? I'm debating which one to try. |
| 16:34.27 | stix | Why do I only see Asterisk18 documentation when going to asterisk.org and Documentation? |
| 16:34.54 | michael-i | It seems whenever Dial() exits non-zero, it jumps right back out of the macro... |
| 16:35.12 | AdvoWork | fullstop, struggling with a phone not registering, SiteA (has asterisk server running trixbox) with multiple phones, all work, site B has phones that connect to siteA.. most work, apart from one, it just wont register :S |
| 16:35.38 | ectospasm | anonymouz666: there is a Digium technician who speaks fluent Spanish, but he currently works night shift, and won't be available for seven hours. |
| 16:36.27 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:37.38 | AdvoWork | logs show: Registrar SIP_EXTENSION@IP timed out |
| 16:37.51 | AdvoWork | even though the other phones use the same IP with no issue :S |
| 16:38.45 | fullstop | AdvoWork: what phone? |
| 16:39.04 | fullstop | With polycom, I've found that you should never ever ever touch the phone through the web interface. |
| 16:39.20 | fullstop | just do everything through a provisioning server and things are much smoother. |
| 16:39.27 | anonymouz666 | ectospasm: thanks for the information ! |
| 16:39.40 | michael-i | My complete extensions.conf (http://pastebin.com/wkPdNJDb) anyone who can help me get an active 'h' extension when one internal phone calls another and hangs up before the other answers gets karma/beer/mentions... |
| 16:40.59 | AdvoWork | fullstop, snom 300 |
| 16:41.54 | AdvoWork | fullstop, the thing is,the phone has been working on/off. If i reset the phone router, i think it will work again, but I can't keep doing that, so trying to work out exactly whats going on |
| 16:43.37 | fullstop | AdvoWork: ppoe and resetting the router reboots the phone? |
| 16:43.44 | fullstop | I do not have any experience with snom phones. |
| 16:48.42 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:48.42 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:50.25 | AdvoWork | it doesnt reset the phone but if i reboot the router, then register, it will more than likely work :S |
| 16:52.52 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 16:53.05 | michael-i | This guy seems to also be seeing what I see...the F option in Dial doesn't work in 1.6? http://lists.digium.com/pipermail/asterisk-users/2010-May/248361.html |
| 16:53.41 | jbroome | so that's be F'd up? |
| 16:53.46 | *** join/#asterisk emc (~cburgdorf@109.231.205.3) |
| 16:53.48 | tuxx- | :-D |
| 16:55.05 | emc | Hi, where can i find independent asterisk consultants in london? (individuals) |
| 16:57.46 | fullstop | I would look in a pub. |
| 16:59.51 | AdvoWork | fullstop, any further suggestions? |
| 17:00.04 | jbroome | even if you don't find one, it's not a wasted trip |
| 17:00.09 | Faustov | ;D |
| 17:00.11 | Faustov | or is it! |
| 17:01.30 | fullstop | AdvoWork: sorry, I don't. Perhaps someone else here is more familiar with snom phones. |
| 17:03.07 | *** join/#asterisk nfi|ermes (~sdff@host54-26-static.93-94-b.business.telecomitalia.it) |
| 17:05.09 | nfi|ermes | FATAL: Module zaphfc not found. |
| 17:05.21 | nfi|ermes | i have not zaphfc module installed |
| 17:05.40 | nfi|ermes | shouldn t it be in dahdi package ? |
| 17:09.03 | fullstop | Actually, I think that you should look for everything in a pub. You never know what one will find at the bottom of a pint. |
| 17:11.51 | nfi|ermes | ? |
| 17:14.07 | fullstop | nfi|ermes: you missed some context from before you joined. |
| 17:19.42 | nfi|ermes | is zaphfc included in asterisk 1.6 or i should install bristuff ? |
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| 17:32.41 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
| 17:33.49 | Letoric | Anybody know an easy way to make asterisk process an array for a gotoif statement, rather than just 1 thing? IE, setting up a handler for obnoxious vendors that treats them differently than normal callers? |
| 17:38.08 | p3nguin_ | Just use one single GotoIf, which sends those calls into a special context with an appropriate dial plan for that type of call. |
| 17:39.21 | Letoric | ok, but with a single gotoif, how do I identify the caller? |
| 17:39.52 | Letoric | that's where I was hoping to put the array in, IE, each of their callerid's gets put into an array and it sends them somewhere if there is a match, else goes to the next line |
| 17:42.18 | robl^laptop | put the numbers in a database.. the check for a match... if it matches, goto annoying-caller context |
| 17:43.56 | Letoric | thanks, but using that method, can I do it in an array, or do I need to run multiple gotoif's? |
| 17:45.57 | robl^laptop | no array or multiple gotoifs needed. you do a quick search in a family (badcallers) using the key Callerid(number). if it returns anything, its a match and then you drop into the correct context |
| 17:46.40 | WindBack | Hello, I have an abnormal problem with some random calls in my Asterisk PBX. Sometimes when I have a call established to the PSTN, it goes down. The internal extension is SIP while the connection to the PSTN is trough DAHDI using ISDN/PRI. Debuging SIP and PRI for some of this calls, I could see the following: http://www.pastebin.ca/1994467. It seems that the PBX is hanging both channels because it is sending a DISCONNECT message to the PSTN and |
| 17:46.40 | WindBack | <PROTECTED> |
| 17:46.52 | Letoric | thanks robl^laptop I'll give it a whirl |
| 17:51.16 | p3nguin_ | letoric: You don't even have to use GotoIf() to pick hte callerid. You can use the old style callerid matching on the extension if you want. |
| 17:53.27 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 17:54.47 | robl^laptop | oldstyle callerID matching is ok for a small number of possible matches.. but if you have more than about 15, I'd go with a db lookup... basically emulate the old blacklist() app |
| 17:58.02 | Letoric | thanks guys. I definitely want the ability to grow beyond a few numbers in there, which is why I was asking about arrays. I'll play with the database suggestion and see if I can pull it together |
| 17:59.23 | robl^laptop | Letoric: look at the DB() function |
| 18:01.39 | robl^laptop | Letoric: GotoIf(${DB_EXISTS(annoying/${CALLERID(number))}?annoyingcallers,s,1) <-- something like that |
| 18:03.05 | thehar | ExecIF($DB_EXISTS(robl/${EXTEN}))?ohmai) |
| 18:03.19 | Letoric | nifty, thanks again |
| 18:04.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:09.28 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 18:20.30 | *** part/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 18:32.36 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:35.31 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 18:35.31 | wcselby | o/ |
| 18:40.55 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 18:47.15 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 18:52.17 | fullstop | I have an IVR which does a Dial(SIP/ip.add.re.ss). When doing sip show channelstats, nothing ever shows up for tx / rx packets for those calls. |
| 18:54.55 | *** join/#asterisk cafer (cafer@88.245.225.149) |
| 18:55.19 | wcselby | what about calls to "Dial(SIP/peername)"? |
| 18:55.42 | *** join/#asterisk luckman212 (~quassel@pool-96-246-172-198.nwrknj.fios.verizon.net) |
| 18:55.56 | fullstop | sorry.. it was a bad description of the dial command |
| 18:56.07 | drift- | [Nov 17 13:50:04] NOTICE[2750]: chan_local.c:504 local_call: No such extension/context 2@users while calling Local channel [Nov 17 13:50:04] NOTICE[2750]: app_dial.c:564 wait_for_answer: Failed to dial on local channel for call forward to '2@users |
| 18:56.09 | drift- | what does that mean? |
| 18:56.21 | fullstop | Dial(SIP/peername@ip.add.re.ss,60,r) |
| 18:56.31 | luckman212 | anyone in here using voip.ms for sip trunking? |
| 18:56.40 | drift- | i am using voip.ms |
| 18:56.45 | cafer | what is problem luckman212 |
| 18:56.58 | luckman212 | drift-: is it true that you cannot do outgoing SIP URI dialing with voip ms? |
| 18:56.59 | wcselby | drift- - you're sending your call into the wrong context, and there is no extension there to handle it |
| 18:57.47 | fullstop | for incoming calls over the ITSP, I get the stats. |
| 18:58.08 | luckman212 | I have tried calling a SIP URI directly via a softphone on my PC, as well as programming a custom extension as SIP://xxx@yyy.com into my dialplan, neither of these methods is working |
| 18:58.20 | luckman212 | I am starting to think its voip.ms |
| 19:00.20 | wcselby | fullstop - hmmm. |
| 19:00.57 | fullstop | yeah, strange |
| 19:00.57 | wcselby | fullstop - I'm not sure I understand, if you have a peername, why do you also have an IP address? Isn't that associated with the peer? |
| 19:01.23 | wcselby | I could understand Dial(SIP/${EXTEN}@ip.add.re.ss,60) |
| 19:01.52 | fullstop | wcselby: that is what I am doing. |
| 19:02.02 | fullstop | Saying peername was incorrect |
| 19:02.08 | wcselby | so this is outbound calls over the ITSP? |
| 19:02.18 | wcselby | is this all outbound calls, or just ones triggered from the IVR? |
| 19:02.29 | fullstop | Inbound calls come in over an ITSP.. |
| 19:02.46 | fullstop | outbound calls are triggered by the IVR and it goes to an external call center. |
| 19:03.12 | fullstop | These go over a lan-to-lan vpn |
| 19:04.12 | *** join/#asterisk vegbox (~kevinle@adsl-64-173-83-182.dsl.lsan03.pacbell.net) |
| 19:04.38 | citywok | what's a good poe switch for 24 phones? (registering as class 1) |
| 19:10.08 | *** join/#asterisk theron (~theron@ip244.scolloc.lh.net) |
| 19:17.42 | wcselby | sorry fullstop - got called away for a bit |
| 19:18.05 | fullstop | not a problem |
| 19:18.24 | wcselby | are you passing off the media stream maybe? |
| 19:18.40 | wcselby | do you have any SIP / RTP traffic after the call has been passed off? |
| 19:19.04 | fullstop | yes, we have sip and rtp traffic |
| 19:19.21 | fullstop | in fact, it's filling the whole pipe |
| 19:19.36 | fullstop | right now, ~80 concurrent calls |
| 19:20.02 | wcselby | i meant, for that particular call |
| 19:20.29 | fullstop | They are all identical calls in that regard |
| 19:21.37 | wcselby | can you show me the CLI output of one of these calls? I realize you just said you had 80 concurrent calls....are all of these calls showing no tx /rx stats? |
| 19:21.43 | fullstop | We receive g729 from the ITSP and pass it right along down to the call center. |
| 19:22.04 | fullstop | No, we have tx / rx stats for calls from the ITSP.. just not down to the call center. |
| 19:22.06 | fullstop | one sec |
| 19:23.00 | fullstop | http://pastebin.com/GBgseB5y |
| 19:23.13 | fullstop | this is from earlier, so it is more like 60 calls |
| 19:24.39 | wcselby | so it's all the calls to 10.21.253.114 ? |
| 19:24.48 | fullstop | correct |
| 19:25.52 | wcselby | that's odd... |
| 19:25.59 | wcselby | i'd almost say file a bug on the issue tracker |
| 19:26.08 | wcselby | what version are you running? |
| 19:26.18 | fullstop | 1.6.2.9 |
| 19:26.27 | fullstop | I know .9 is a wee bit old |
| 19:27.22 | wcselby | what command are you using to generate that output? |
| 19:27.36 | *** join/#asterisk RAFAIP (~chatzilla@190.145.253.51) |
| 19:27.38 | fullstop | sip show channelstats |
| 19:27.42 | wcselby | heh |
| 19:27.44 | wcselby | of course |
| 19:27.54 | wcselby | must not have been used on 1.4.x |
| 19:28.00 | fullstop | ? |
| 19:28.01 | wcselby | let me load up one of my 1.6.2.x boxes |
| 19:28.07 | fullstop | ahh |
| 19:28.09 | fullstop | i see |
| 19:31.52 | wcselby | of course, no one is making any calls on that box today |
| 19:31.52 | wcselby | lol |
| 19:31.58 | fullstop | haha |
| 19:32.16 | wcselby | ah well, I wish I could hlep you out more, but it's not something I'm able to reproduce. I'd suggest submitting a bug and letting them take a look |
| 19:32.45 | fullstop | I've been pretty lucky at finding bugs since I started using Asterisk.. |
| 19:32.49 | WIMPy | sings: No calls today. My Ast'risk's gone away. |
| 19:33.23 | wcselby | fullstop - heh |
| 19:33.32 | wcselby | for a call center that size, I can understand how |
| 19:33.50 | wcselby | my larger installs have about 30-35 concurrent calls at peak times |
| 19:33.55 | fullstop | overall it has been a good experience. |
| 19:34.06 | fullstop | They are expanding.. possibly > 100 concurrent soon. |
| 19:34.19 | wcselby | my smaller ones (like that 1.6.2.x box I just talked about) have maybe 5 people in the office total |
| 19:34.30 | wcselby | fullstop - do you use SIP trunks or PRI lines? |
| 19:34.41 | fullstop | sip trunk for the inbound calls |
| 19:34.54 | *** join/#asterisk kryl (~kryl@aqu33-2-82-224-109-232.fbx.proxad.net) |
| 19:34.56 | fullstop | outbound goes over a VPN |
| 19:34.58 | kryl | hi |
| 19:35.07 | fullstop | over a plain vanilla bonded T |
| 19:35.20 | wcselby | how many inbound sip trunks do you have? |
| 19:35.33 | fullstop | just the one, over mpls |
| 19:35.47 | wcselby | who is your provider? |
| 19:35.58 | fullstop | we are being billed for 96 concurrent calls.... but it tops out at 80 and callers get a busy signal. |
| 19:35.59 | wcselby | when I think of sip trunk, I think of concurrent calls (I know sip trunk is a misnomer) |
| 19:36.04 | kryl | I'm looking for a way to link iphone/android clients with sip account (3g), in order to create call meeting ! do you think asterisk can help me for that ? |
| 19:36.08 | fullstop | PAETEC |
| 19:36.20 | kryl | is there a way to crypt the voice data ? (optional) |
| 19:36.37 | wcselby | kryl - use a sip client on each phone, then setup an asterisk server that they register to, then give them an extension for a meetme conference room |
| 19:39.16 | wcselby | fullstop - interesting. haven't heard of them. what's your typical monthly bill for 96 concurrent calls? |
| 19:39.26 | wcselby | or do you pay by the minute, etc? |
| 19:39.54 | fullstop | wcselby: I'm not part of that discussion. :D |
| 19:39.59 | wcselby | heh |
| 19:40.00 | wcselby | gotcha |
| 19:40.02 | fullstop | wcselby: I honestly don't know. |
| 19:40.05 | wcselby | just curious |
| 19:40.30 | fullstop | It's not by the minute. We rack up a lot of minutes. |
| 19:40.55 | fullstop | 506,000 minutes last month |
| 19:41.02 | wcselby | it's always been my experience that anything over 18-22 concurrent calls, it tended to be cheaper to go the PRI route, but I've mostly dealt with SIP trunk providers like bandwidth.com, broadvox, etc. |
| 19:41.32 | kryl | wcselby, I'm beginner to asterisk world ! |
| 19:41.41 | wcselby | kryl - check out the book |
| 19:41.43 | wcselby | ~book |
| 19:41.43 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 19:42.17 | kryl | excellent, thank you :) |
| 19:50.29 | *** join/#asterisk marl_scot (~matt.lowe@office.unk.com) |
| 19:53.06 | *** join/#asterisk CrashSys (~james@rrcs-97-76-33-146.se.biz.rr.com) |
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| 19:59.18 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:05.46 | *** join/#asterisk JoeT (~None@cpe-173-168-180-115.tampabay.res.rr.com) |
| 20:07.31 | *** join/#asterisk gehko_ (~mint@173-162-245-205-NewEngland.hfc.comcastbusiness.net) |
| 20:07.54 | JoeT | HI folks... Quick question. (Asterisk 1.8). When a user on a SIP phone lifts the handset and dials an extension, Asterisk takes 10 seconds to process the extension even if there is no ambiguity in the match (i.e. the context has only 1 digit extensions and no pattern matching). If the user dials the extension and then picks up the handset, the extension is dialed immediately. I can't |
| 20:07.54 | JoeT | figure out how to correct this. Can anyone? |
| 20:08.40 | *** join/#asterisk simplydrew (~simplydre@96.238.59.82) |
| 20:09.21 | gehko_ | Has anyone gotten dialogic cards working with asterisk? |
| 20:09.23 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:09.30 | jaytee | ~book |
| 20:09.30 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 20:10.26 | WIMPy | JoeT: That's most probably a phone issue. |
| 20:11.19 | WIMPy | gehko_: That would have to be via CAPI. Not sure how good that option is nowadays. |
| 20:11.42 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 20:11.44 | gehko_ | is mISDN better? |
| 20:11.47 | *** join/#asterisk r0d3nt (~astrutt@cheshire.telephreak.org) |
| 20:12.28 | WIMPy | It surely looks VERY promising in 1.8, but I haven't tried it, yet. I'm using misdn2 with lcr. |
| 20:13.07 | WIMPy | (that is the misdn that comes in the standard kernel) |
| 20:13.08 | wcselby | JoeT - what type of phone is this issue occuring on? |
| 20:13.54 | JoeT | Sipura SP-841 |
| 20:15.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 20:17.02 | wcselby | JoeT - you need to set the "Dial Plan" in that phone to be less than 10 seconds |
| 20:17.54 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 20:18.24 | JoeT | wcselby: Do you know where? I don't see anything similar in the configuration. |
| 20:18.49 | wcselby | i think you need to be in the admin section of the config website, then go to the line config and find the Dial Plan option |
| 20:19.02 | wcselby | i was just looking at a pdf with the specifics |
| 20:19.05 | gehko_ | i am having a very difficult time with mISDN in centos |
| 20:19.12 | wcselby | google for "sipura sp-841 dial plan" |
| 20:19.12 | timholum | I need to write a web interface to be able to transfer calls ext... does anyone know of a good asterisk manager interface library to use, I would prefer php but it could also be in python or ruby |
| 20:19.15 | wcselby | and you'll see it |
| 20:19.32 | vegbox | Anyone here use XO IP Flex service? |
| 20:19.55 | WIMPy | gehko_: misdn1 or misdn2? |
| 20:20.02 | WindBack | Hello, I have an abnormal problem with some random calls in my Asterisk PBX. Sometimes when I have a call established to the PSTN, it goes down. The internal extension is SIP while the connection to the PSTN is trough DAHDI using ISDN/PRI. Debuging SIP and PRI for some of this calls, I could see the following: http://www.pastebin.ca/1994467. It seems that the PBX is hanging both channels because it is sending a DISCONNECT message to the PSTN and |
| 20:20.02 | WindBack | <PROTECTED> |
| 20:20.08 | gehko_ | WIMPy: misdn1 |
| 20:20.12 | JoeT | OK, found the dial plan.. Will google it and see.. |
| 20:21.00 | WIMPy | gehko_: The original misdn1 or the fork for th Asterisk 1.8 chan_misdn? |
| 20:21.24 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 20:21.48 | WIMPy | gehko_: It can be a bit tricky to get the right version of misdn for your kernel version. That's why I gave up some time and changed to misdn2. |
| 20:22.00 | gehko_ | WIMPy: the fork for asterisk 1.6 |
| 20:22.13 | tzafrir_laptop | WindBack, so why does asterisk decide to hang up? |
| 20:22.24 | WIMPy | Ups. There was another fork? |
| 20:22.37 | WindBack | tzafrir_laptop: this is my question |
| 20:22.49 | gehko_ | WIMPy: there is one for 1.4, 1.6 and 1.8 |
| 20:23.04 | WIMPy | gehko_: I mean of the kernel modules. |
| 20:23.13 | Katty | hai |
| 20:23.25 | WindBack | tzafrir_laptop: this is the my doubt, I don't know why it is doing so |
| 20:23.28 | WIMPy | gehko_: Do you only need TE or NT mode as well? |
| 20:23.43 | tzafrir_laptop | It seems that the trace you show begins slightly after that |
| 20:23.54 | tzafrir_laptop | Asterisk has already decided to hang up, right? |
| 20:24.39 | gehko_ | WIMPy: just te (for now :) |
| 20:25.17 | WindBack | tzafrir_laptop: the call was established 10 minutes before. And sudenly it hangs the call. The trace of the hang starts where I showed you in pastebin. There is nothing before |
| 20:25.39 | tzafrir_laptop | WindBack, is it reproducable? |
| 20:25.55 | WIMPy | gehko_: Well, think about that "for now" bit then. misdn1 has a reputation of being unstable. In myu experience only if you use NT mode, however. |
| 20:25.59 | WindBack | tzafrir_laptop: it is a very random bahviour |
| 20:26.22 | WindBack | tzafrir_laptop: it not happens always |
| 20:26.35 | tzafrir_laptop | WindBack, next thing I would do would be enabling more debug |
| 20:26.39 | WindBack | tzafrir_laptop: it was not easy to capture this log |
| 20:26.42 | JoeT | wcselby: That fixed it. Thank you! |
| 20:26.45 | tzafrir_laptop | not sip debug and pri debug at first stage |
| 20:27.05 | WindBack | tzafrir_laptop: what kind of debug i can enable_ |
| 20:27.07 | WindBack | ? |
| 20:27.25 | tzafrir_laptop | and try to figure out "where the hangup began". This spans multiple channels (the legs of the call) |
| 20:27.38 | gehko_ | WIMPy: i dont think we are going to need it, i have SIP working with te mode now |
| 20:27.59 | gehko_ | WIMPy: but that is through the diva SIPcontrol interface |
| 20:28.28 | WIMPy | gehko_: Err, what? |
| 20:28.30 | gehko_ | WIMPy: i wanted to get asterisk in the mix as well, but it is trying my patience |
| 20:29.09 | WindBack | tzafrir_laptop: what kind of debug do you think can I enable to see more details when the hang begin? |
| 20:30.23 | tzafrir_laptop | core set debug 2 |
| 20:30.29 | tzafrir_laptop | or maybe 3 or 4 |
| 20:31.02 | WindBack | tzafrir_laptop: ok, thanks for the recomendation |
| 20:31.40 | WIMPy | gehko_: I think for a new install I'd try Asterisk 1.8 with dahdi or chan_misdn and the coresponding driver fork, depending on the hardware. But with a diva i'm pretty sure, capi is your only option. |
| 20:31.55 | WindBack | tzafrir_laptop: do you think this could be a bug in *? I'm using 1.6.1.18 and dahdi 2.3 |
| 20:32.23 | gehko_ | i just noticed that eicon has a capi config in there driver install |
| 20:32.32 | gehko_ | WIMPy: ^^ |
| 20:33.43 | tzafrir_laptop | WindBack, no idea |
| 20:33.58 | WindBack | tzafrir_laptop: ok, thanks |
| 20:35.38 | jaytee | ~itsp-uslist |
| 20:35.57 | jaytee | ~itspus-list |
| 20:35.59 | p3nguin_ | ~itsplist-us |
| 20:35.59 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 20:36.11 | jaytee | thanks, it's been awhile :-) |
| 20:41.35 | jaytee | anyone here used Vonage softphone account as a "SIP trunk"? |
| 20:47.08 | *** join/#asterisk slacker775 (~dhollis@static-96-254-30-130.tampfl.fios.verizon.net) |
| 20:48.44 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-075-154-214.pools.arcor-ip.net) |
| 20:50.08 | slacker775 | is there an easy way to set a dahdi channel to busy/offhook? I have a line that doesnt seem to be working right and would like to take it out of service so callers arent hitting it and having issues |
| 20:54.29 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:54.29 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.4 (2010/09/01) -=- Visit the new official Asterisk wiki: wiki.asterisk.org |
| 21:02.35 | gehko_ | has anyone found the 'Configure hardware' or 'mISDN Config' sections of the GUI? |
| 21:03.23 | WIMPy | gehko_: What GUI? And are you sure, there is one? |
| 21:04.18 | gehko_ | WIMPy: i am trying to add a BRI Trunk to asterisk and that is what asteriskNOW keeps telling me, but i cant locate it anywhere |
| 21:04.40 | p3nguin_ | gehko_: #AsteriskNOW |
| 21:04.46 | wcselby | gehko_ - which GUI are you using? |
| 21:05.04 | gehko_ | wcselby: web |
| 21:05.17 | wcselby | gehko_ - which GUI are you using? there's more than one web gui with AsteriskNOW |
| 21:05.55 | gehko_ | wcselby: the one that installs with the asterisk linux iso |
| 21:06.00 | wcselby | ..... |
| 21:06.05 | gehko_ | ??? |
| 21:06.08 | gehko_ | sorry |
| 21:06.11 | p3nguin_ | There were choices. |
| 21:06.34 | p3nguin_ | And with either choice, this is the wrong place. |
| 21:06.55 | gehko_ | Asterisk GUI-version : SVN--rexported |
| 21:07.21 | gehko_ | p3nguin_: :) i knew you were going there |
| 21:07.52 | wcselby | then I'm sorry I've only used that GUI once, you'll have to ask in one of the other channels |
| 21:08.45 | gehko_ | i dont care to use the gui. i just want to add a bri trunk to asterisk |
| 21:09.35 | p3nguin_ | If you add it to Asterisk without the GUI, there's a chance it will be wiped out the very next time you apply a change with the GUI. |
| 21:10.41 | gehko_ | p3nguin_: i am fine with that because the GUI has been pretty useless for me |
| 21:11.59 | WIMPy | gehko_: Then you might be better off, to install a 'normal' plain Asterisk. |
| 21:12.46 | p3nguin_ | AsteriskNOW offers, in addition to the GUI choice, the option to install Asterisk without any GUI. |
| 21:13.44 | gehko_ | i hate to do a reinstall, but it looks like i am going to have to. |
| 21:14.05 | p3nguin_ | You don't have to. |
| 21:14.11 | p3nguin_ | Just uninstall the GUI. |
| 21:14.20 | gehko_ | with yum? |
| 21:14.23 | p3nguin_ | yes |
| 21:14.32 | p3nguin_ | or even rpm. |
| 21:15.18 | p3nguin_ | Asterisk may need a couple settings changed so it isn't expecting the GUI to be present anymore, but that's going to be quite trivial. |
| 21:16.06 | WIMPy | Hmm. Does that clean the configuration? |
| 21:16.18 | *** join/#asterisk xibalba (~reza@216.105.40.8) |
| 21:16.38 | xibalba | hello everyone, can someone help point me to configuring a sip trunk ip 2 ip |
| 21:16.46 | xibalba | i dont want to do any registration |
| 21:17.25 | gehko_ | i have asterisk-gui.noarch listed as installed. is that the only one i need to erase? |
| 21:18.25 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.72) |
| 21:19.53 | leifmadsen | xibalba: host=xxx.xxx.xxx.xxx |
| 21:20.32 | gehko_ | ok, now that i have the gui uninstalled how do i add the bri trunk? |
| 21:21.19 | WIMPy | gehko_: You've got capi running? |
| 21:21.31 | gehko_ | yes |
| 21:21.42 | xibalba | leifmadsen ; if that is the only thing i put it will no attempt to register? |
| 21:21.52 | WIMPy | Then chan_capi.conf is your friend. |
| 21:22.06 | leifmadsen | xibalba: the [peer] section never tries to register. Only the register => lines do that. |
| 21:22.20 | leifmadsen | xibalba: remove the register => lines for things you don't want Asterisk to register to. |
| 21:22.34 | xibalba | hmm, i may be mistekn here because i'm doing this through a freepbx portal which is based on asterisk |
| 21:22.42 | leifmadsen | xibalba: host=dynamic is used when the other peer is registering to your asterisk server. If they don't register, then you need to use the host=x.x.x.x (IP address) |
| 21:23.02 | leifmadsen | xibalba: sorry, I can't help you with freepbx. I'd suggest asking over in #freepbx |
| 21:23.48 | xibalba | ok thanks for the info thus far |
| 21:23.49 | xibalba | see ya |
| 21:23.50 | *** part/#asterisk xibalba (~reza@216.105.40.8) |
| 21:25.54 | gehko_ | WIMPy: i have a capi.conf? |
| 21:27.45 | WIMPy | It's been quite soe years I tried to use capi. But the capi itself sould have a capi.conf IIRC. And Asterisk has a chan_capi.conf. |
| 21:27.45 | gehko_ | WIMPy: i guess i am not understanding how to tell asterisk to use capi or setup a bri trunk with a capi device |
| 21:28.54 | *** join/#asterisk MiserySoft (~Lee@nat76.mia.three.co.uk) |
| 21:30.02 | WIMPy | starts to wonder if chan_capi actually has been part of Asterisk or if it was seperate. |
| 21:31.11 | WIMPy | No, seems to be a seperate thing. |
| 21:31.33 | WIMPy | google for chan_capi and you will find a howto inthe forst hits. |
| 21:31.55 | WIMPy | Hmm/ |
| 21:32.16 | WIMPy | Maybe it would be a good idea to write a Asterisk vs ISDN howto or something. |
| 21:36.24 | gehko_ | yah i found it.. |
| 21:36.27 | gehko_ | its working now |
| 21:38.49 | gehko_ | hah |
| 21:39.14 | gehko_ | its funny...it was probably working the whole time, i just didnt know what the f any of this was |
| 21:46.51 | gehko_ | WIMPy: how would i go about dialing something to try it out. the page gives: s,1,Dial,CAPI/${MSN}:b${EXTEN}|30 |
| 21:47.02 | gehko_ | WIMPy: but i dont know what to do with that string |
| 21:47.34 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:48.51 | p3nguin_ | First you need to translate it into proper syntax. s,1,Dial(CAPI/${MSN}:b${EXTEN},30) |
| 21:49.00 | gehko_ | it seems i need to find Dial() |
| 21:49.24 | WIMPy | That goes into extensions.conf. |
| 21:49.32 | p3nguin_ | What do you mean find it? It's an application. |
| 21:50.15 | gehko_ | yar, but it is not on my system |
| 21:50.21 | WIMPy | gehko_: It might be a good idea to get some more overview with the |
| 21:50.24 | WIMPy | ~book |
| 21:50.24 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 21:50.26 | drift- | <PROTECTED> |
| 21:50.28 | drift- | what does that mean? |
| 21:50.30 | drift- | where should i check? |
| 21:50.32 | p3nguin_ | What makes you think it's not an application? |
| 21:50.57 | gehko_ | i didnt say it wasnt an application |
| 21:51.17 | p3nguin_ | What makes you think it doesn't exist on your system? |
| 21:51.49 | gehko_ | which Dial does not yeild results so therefore i must 'find' it |
| 21:52.04 | p3nguin_ | It's not a command used in the Linux shell. |
| 21:52.09 | jbroome | it's no a linux command, it's an asterisk command |
| 21:52.12 | WIMPy | That syntax seems to be outdated. And as even voip-info calls it outdated, it must be very outdated. |
| 21:52.14 | p3nguin_ | It's an internal Asterisk application. |
| 21:52.35 | gehko_ | p3nguin_: ah |
| 21:53.04 | gehko_ | "the more you know" shooting star just flew over my computer |
| 21:53.09 | p3nguin_ | /usr/lib/asterisk/modules/app_dial.so |
| 21:53.33 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:54.02 | WIMPy | CAPI/g<group>/dest CAPI/contr<card>/dest or CAPI/<interface>/dest + optional /<options> |
| 21:54.07 | drift- | how can i fix this error ? [Nov 17 16:44:25] NOTICE[3173]: chan_local.c:504 local_call: No such extension/context 2@users while calling Local channel [Nov 17 16:44:25] NOTICE[3173]: app_dial.c:564 wait_for_answer: Failed to dial on local channel for call forward to '2@users' ? doesnt make sense :( |
| 21:54.35 | p3nguin_ | 2@users does not exist. What part of that doesn't make sense? |
| 21:54.39 | jbroome | drift-: i think a repost every 5 mins is when you get help. you just missed it |
| 21:54.40 | WIMPy | drift-: Don't dial non-existing extensions. |
| 21:55.28 | drift- | that extension exists tho |
| 21:55.33 | drift- | 205 is working extension |
| 21:55.48 | p3nguin_ | Show us that extension 2 exists in [users]. |
| 21:55.49 | drift- | 205/205 172.16.1.28 D 5060 Unmonitored |
| 21:55.56 | russellb | '2', not '205' |
| 21:56.02 | drift- | i dial 205 |
| 21:56.03 | russellb | and that's not an extension, that's a SIP peer |
| 21:56.04 | p3nguin_ | We don't need to see SIP/205. |
| 21:56.10 | p3nguin_ | Show us that extension 2 exists in [users]. |
| 21:56.23 | drift- | i am dialing 205 and send |
| 21:56.34 | jbroome | drift-: are you reading your own error? |
| 21:56.34 | russellb | then your dialplan is broken |
| 21:56.55 | russellb | you should have something like, exten => 205,1,Verbose(1,Hello World) |
| 21:57.29 | russellb | I would recommend reviewing your dialplan and convincing yourself which extension the call should be matching |
| 21:57.47 | p3nguin_ | But then you have to Dial(Local/205@users) instead of 2@users like it says. |
| 21:58.11 | drift- | every other extension works except this one |
| 21:58.35 | wcselby | drift- - are you using a web gui? |
| 21:58.44 | wcselby | drift- - or was this configured in extensions.conf ? |
| 21:58.58 | drift- | no custom |
| 21:59.00 | drift- | i have 2 broken extensions |
| 21:59.04 | drift- | out of like the 10 i got |
| 21:59.08 | drift- | extension 202 and 205 |
| 21:59.22 | p3nguin_ | Are you going to show us or just keep complaining about it? |
| 21:59.26 | wcselby | drift- - then please pastebin your extensions.conf [users] context |
| 21:59.30 | drift- | ok 1 sec |
| 21:59.58 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 22:00.38 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 22:01.36 | drift- | http://pastebin.com/hqTg1jC9 |
| 22:01.39 | drift- | my extensions.conf |
| 22:02.27 | p3nguin_ | As the error shows, extension 2 does not exist in the users context. |
| 22:02.39 | drift- | i'm trying to dial 205 and 202 |
| 22:03.12 | p3nguin_ | How? What are you doing to Dial those numbers? |
| 22:03.22 | wcselby | drift- - no you're not, you're dialing 2@users |
| 22:03.23 | drift- | just typing 205 and send |
| 22:03.28 | drift- | no i am not |
| 22:03.29 | wcselby | typing 205 in what |
| 22:03.33 | drift- | my phone |
| 22:03.35 | p3nguin_ | What context is your device in? |
| 22:03.35 | wcselby | what phone |
| 22:03.36 | drift- | i pick up phone and dial 205 |
| 22:03.41 | drift- | my extension 211 |
| 22:03.44 | fullstop | drift-: dial 205 then pick up the phone |
| 22:03.45 | wcselby | what type of phone |
| 22:03.47 | p3nguin_ | What context is your device in? <----- |
| 22:04.09 | fullstop | your phone probably has a built in digit map, and it is sending the digits early. |
| 22:04.17 | wcselby | what fullstop said |
| 22:04.17 | fullstop | alter the digit map and you should be good to go. |
| 22:04.25 | wcselby | digitmap / dial plan |
| 22:04.31 | drift- | but it was just working... |
| 22:04.32 | russellb | actually, the question is where does the Local channel come in ... |
| 22:04.35 | drift- | and works for 10 other extensions |
| 22:04.35 | fullstop | or just get used to dialing before picking up the phone. :D |
| 22:04.42 | russellb | I don't see chan_local in that dialplan, but that's where the error comes from |
| 22:05.02 | p3nguin_ | russellb: We may never know, since he won't bother telling us any useful information. |
| 22:05.12 | *** join/#asterisk candrews (~candrews@fsf/member/candrews) |
| 22:05.13 | russellb | be nice:-) |
| 22:05.15 | p3nguin_ | On that note, I'm done. |
| 22:05.18 | wcselby | haha p3nguin_ and russellb |
| 22:05.30 | drift- | fullstop it goes busy |
| 22:05.32 | wcselby | his macro-phone is a little odd too |
| 22:05.34 | drift- | i dial 205 and pick up |
| 22:05.41 | wcselby | drift- - what type of phone? |
| 22:05.53 | drift- | polycom 501 |
| 22:06.11 | wcselby | check your digitmap |
| 22:06.47 | wcselby | and....what context is your phone assigned to |
| 22:06.55 | WIMPy | Well, THAT error certainly does not come from THAT dialplan. So what is missing? |
| 22:07.01 | wcselby | why are you getting a local channel in the error |
| 22:07.21 | drift- | [211] type=friend host=dynamic secret=12345 context=users |
| 22:07.31 | drift- | [205] type=friend host=dynamic secret=12345 context=users2 |
| 22:07.42 | russellb | unless a phone has been set up to do a call forward to '2' |
| 22:07.46 | russellb | that would do it, heh |
| 22:07.52 | wcselby | russellb - lol yeah that's true |
| 22:07.52 | drift- | nah because |
| 22:08.00 | drift- | i just tried diffrent phones |
| 22:08.04 | wcselby | drift- - check the peer 205 |
| 22:08.15 | russellb | right, check the phone you're trying to call, and see if it is set to forward |
| 22:08.18 | Katty | drmessano: ping. |
| 22:08.26 | Katty | also! |
| 22:08.31 | Katty | heeeelllllllllllllllloooooooo!!! |
| 22:08.34 | Katty | hugs WIMPy |
| 22:08.36 | russellb | hi2u2 Katty |
| 22:08.38 | Katty | hugs wcselby |
| 22:08.41 | Katty | hugs russellb |
| 22:08.42 | fullstop | Not the hugs again |
| 22:08.46 | WIMPy | russellb: Uless thats done on the dialplan, how would that local channel come from? |
| 22:08.47 | russellb | oh yes, the hugs. |
| 22:08.47 | Katty | yes. the hugs. |
| 22:08.48 | Katty | again. |
| 22:08.48 | wcselby | o/ Katty |
| 22:08.51 | Katty | hugs fullstop |
| 22:08.56 | fullstop | and, to bring up the bad pun from the other day.. |
| 22:08.57 | russellb | WIMPy: local channel is used internally to process a call forward |
| 22:08.59 | russellb | it's magic |
| 22:09.02 | wcselby | WIMPy - a 183 Redirect or whatever the number is |
| 22:09.02 | fullstop | At least most everyone here uses Dial() |
| 22:09.07 | drift- | i just went to 205 and called my self |
| 22:09.08 | drift- | it works fine |
| 22:09.13 | wcselby | drift- - NO |
| 22:09.26 | wcselby | drift- - look at the phone 205 and see if it's call forwarding enabled |
| 22:09.31 | *** join/#asterisk tecnico (~tecnico@75.76.169.148) |
| 22:10.02 | wcselby | either that, or bump up verbosity and show us the CLI of the entire call, not just the error |
| 22:10.03 | p3nguin_ | CallForwardAll -> 2 |
| 22:10.04 | drift- | i dont even know where that is heh hang on |
| 22:10.04 | p3nguin_ | heh |
| 22:10.44 | wcselby | p3nguin_ - i've seen people do it |
| 22:10.44 | Katty | hugs p3nguin_ |
| 22:10.50 | wcselby | they're checking out the settings on their phones |
| 22:11.06 | wcselby | stumble into that, and try hitting 1 or 2 to back out before realizing what to do |
| 22:11.08 | p3nguin_ | Big Red Button syndrome? |
| 22:11.08 | drift- | omg |
| 22:11.10 | *** join/#asterisk cusco (~trilili@57.218.108.93.rev.vodafone.pt) |
| 22:11.11 | drift- | somone did forward |
| 22:11.12 | drift- | phone 2 |
| 22:11.14 | cusco | hi |
| 22:11.15 | drift- | what retards |
| 22:11.15 | jbroome | HAHAHA |
| 22:11.17 | drift- | LOL |
| 22:11.29 | russellb | WINS! |
| 22:11.34 | *** join/#asterisk Dovid (~Dovid@173.220.127.18) |
| 22:11.43 | jbroome | the president of my company tends to do that, then go in a tizzy because all her incoming calls are going to vm |
| 22:11.45 | wcselby | russellb - :) I'd have gotten there with the full CLI log heh |
| 22:11.47 | russellb | I expect high fives from everyone for calling it |
| 22:11.53 | jbroome | ^5 |
| 22:11.58 | russellb | \o/ |
| 22:12.07 | cusco | peer answers a call from queue and wishes to press *00 so the call gets transfered to another pstn number via dahdi |
| 22:12.14 | cusco | what is the best way to acomplish this? |
| 22:12.16 | cusco | features.conf? |
| 22:12.36 | WIMPy | hands the Asterisk expert of the day badge to russellb |
| 22:12.38 | drift- | lol thanks guys |
| 22:12.39 | drift- | :) |
| 22:12.43 | drift- | i'm flippin out thinkikn i dint touch shit all day |
| 22:12.46 | drift- | how hell can phones just break |
| 22:13.12 | WIMPy | drift-: Disable that feature on the phone. |
| 22:13.34 | gehko_ | do i have to restart the asterisk service when i make a change in extensions.conf? |
| 22:13.43 | cusco | say featurename => *00,caller,Dial(bla) |
| 22:13.43 | p3nguin_ | gehko_: No, dialplan reload is enough. |
| 22:13.46 | cusco | should work, right? |
| 22:13.48 | WIMPy | gehko_: Dialplan reload |
| 22:14.12 | gehko_ | p3nguin_: WIMPy : thanks for all your help today |
| 22:14.17 | p3nguin_ | gehko_: That's run from the Asterisk CLI, if you didn't know. |
| 22:14.20 | timholum | can someone point me in the correct direction where I can learn the Asterisk Manager API comand to transfer a call? |
| 22:14.45 | gehko_ | p3nguin_: got it, thanks :) |
| 22:15.16 | frigidzephyr | timholum: https://wiki.asterisk.org/wiki/display/AST/AMI+Actions |
| 22:15.45 | timholum | thanks frigidzephyr |
| 22:15.58 | *** join/#asterisk RAFAIP (~chatzilla@190.145.253.51) |
| 22:16.35 | RAFAIP | hi |
| 22:16.41 | timholum | I am defenetly going to have to bookmark that site |
| 22:17.48 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 22:18.02 | *** join/#asterisk tecnico (~tecnico@75.76.169.148) |
| 22:18.51 | RAFAIP | the Action: SetVar in asterisk manager, the variable I set is of what type? global? |
| 22:19.19 | *** part/#asterisk slacker775 (~dhollis@static-96-254-30-130.tampfl.fios.verizon.net) |
| 22:22.51 | candrews | I have multiple gtalk accounts set up in gtalk.conf. How do I setup extensions.conf to route calls to a specific gtalk accounts to a SIP number? |
| 22:23.03 | candrews | I tried this, but it didn't work: |
| 22:23.03 | candrews | exten => candrewsintegralblue@gmail.com,n,Dial(SIP/100,20) |
| 22:24.37 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:27.43 | *** join/#asterisk mun24 (~chatzilla@mail.soti.net) |
| 22:28.14 | mun24 | I am running asterisk on ubuntu. I am getting this error |
| 22:28.33 | mun24 | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 22:28.36 | thehar | out |
| 22:29.45 | WIMPy | Actially I guess your're actually not running it. Try starting it with asterisk -cvvv and see what happens. |
| 22:31.38 | mun24 | I can see the files in the /var/run/asterisk/asterisk.ctl |
| 22:32.07 | mun24 | It may be the permission issue, whcih I don't know how to fix it |
| 22:36.04 | ChannelZ | candrews: I haven't played with gtalk in awhile but don't you just drive them into separate contexts? |
| 22:36.30 | candrews | I can't figure out how to dump the different accounts calls into different contexts in gtalk.conf |
| 22:37.42 | cusco | hey... |
| 22:37.44 | ChannelZ | context=xxxxxx in each [whoever] ? |
| 22:37.45 | cusco | chan_sip.c:17946 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '3be1c6ae5ffc741723c79256295041db@10.100.100.5'. Giving up. |
| 22:37.50 | cusco | how can I diagnose that? |
| 22:38.09 | WIMPy | Hm. After the forwarding trick, I took a look into DND and found it to be one of the examples, where HANGUPCAUSE doesn't make sense. |
| 22:40.06 | RAFAIP | the Action: SetVar in asterisk manager, the variable I set is of what type? global? |
| 22:40.58 | RAFAIP | how can I set a variable for a call I receive? |
| 22:41.20 | ChannelZ | it says Channel variable |
| 22:43.11 | *** join/#asterisk moy_ (~moy@200.7.206.126) |
| 22:43.16 | RAFAIP | thats right, but I need to set this variable for the channel who is calling me, because in the context I have a exten=>h,1,goto(prueba,s,1) and in the context prueba i need to read this variable |
| 22:44.03 | *** join/#asterisk jetlag (~jetlag@pool-173-61-243-204.cmdnnj.east.verizon.net) |
| 22:44.51 | RAFAIP | i mean, when I hang up, who is calling me is sent to this context, but I need to set a variable that is going to be used in that context, and I need to do it from a web application, i'm using AMI for it |
| 22:46.24 | ChannelZ | Channel variables belong to the channel they are set on. If you want global variables, use GLOBAL, or the AstDB or something. But you have to manage that space yourself (multiple calls overwriting the same global var...) |
| 22:47.33 | RAFAIP | I dont want to use a global variable, thats my trouble |
| 22:50.01 | ChannelZ | Well as opposed to what? You want to pass information between two unrelated things so you have to do it by *some* means they have in common. |
| 22:50.53 | ChannelZ | (IE you have Channel A with a piece of information you want Channel B to act upon when they hang up?) |
| 22:51.19 | Kobaz | RAFAIP: SHARED() |
| 22:51.25 | Kobaz | RAFAIP: or use my group variables patch |
| 22:51.48 | ChannelZ | Ah interesting. Didn't know about shared. |
| 22:51.55 | Kobaz | RAFAIP: you don't need to private message me |
| 22:52.00 | RAFAIP | ok |
| 22:52.12 | ChannelZ | Let me guess, you're building a robo-dialer |
| 22:52.41 | *** join/#asterisk tyman (~tyler@173-14-203-39-fresno.hfc.comcastbusiness.net) |
| 22:52.41 | Kobaz | you can share data between channels with SHARED |
| 22:52.53 | Kobaz | but... you need to know the channel name of the channel you want to share data with |
| 22:53.04 | RAFAIP | that my big trouble |
| 22:53.25 | RAFAIP | how can i know the channel of the call i'm receiving |
| 22:54.54 | Kobaz | ${CHANNEL} |
| 22:55.13 | RAFAIP | but how i do it from the ami? |
| 22:55.28 | RAFAIP | let me explain you what i'm building |
| 22:55.33 | Kobaz | a bit |
| 22:56.13 | gehko_ | where would ael-demo config info be coming from if it is not in extenstions.conf |
| 22:57.07 | gehko_ | ^? |
| 22:58.19 | RAFAIP | i have a web application in wich i make a query to a database, and i need to pass the result of it to who calls me, so, i know my extension adn my own channel, but i dont know the channel of who is calling me, so, i cant set a variable for that channel |
| 22:58.31 | ChannelZ | extensions.ael |
| 22:59.12 | ChannelZ | if you're loading that module and it's finding it perhaps. Or if you have something else #included from your extensions.conf. The possibilities are many |
| 22:59.43 | citywok | RAFAIP: why don't you have the dialplan make the call from the channel for which you need to set the variable? |
| 22:59.45 | WIMPy | RAFAIP: Watch out for a bridge message on ami. |
| 23:00.03 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:00.25 | gehko_ | ChannelZ: i renamed my extensions.ael and my extenstions.conf has no reference to it |
| 23:00.29 | citywok | or as WIMPy suggested, if you know one leg of the channel, as soon as the channel's are bridged itw ill go by in the AMI, and you can set it. |
| 23:00.39 | ChannelZ | gehko_: do 'module show like pbx_ael' |
| 23:01.33 | gehko_ | Module Description Use Count |
| 23:01.33 | gehko_ | pbx_ael.so Asterisk Extension Language Compiler 0 |
| 23:01.49 | ChannelZ | actually.. if you're seeing it in 'dialplan show', it should tell you where it came from... "[ Context 'out_local' created by 'pbx_config' ]" for instance |
| 23:02.14 | RAFAIP | bridged mesage on ami? and where i can find info about it? |
| 23:02.40 | citywok | telnet in to the ami, enable logging to disk |
| 23:02.40 | Nugget | telnet is eeeeeeevil! |
| 23:02.50 | citywok | make an example call, and then look at the result, you'll be able to read it pretty quickly. |
| 23:06.39 | RAFAIP | i can get the channel of the call I receive from the dialplan? |
| 23:08.43 | gehko_ | ChannelZ: i had to restart the asterisk service for it to unload i thought i could just dialplan reload |
| 23:09.39 | citywok | stop PMing |
| 23:09.52 | RAFAIP | ? |
| 23:10.04 | mun24 | I have asterisk process running and it process files in /var/run/asterisk folder, I am unable to connect to these process files, getting this error Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 23:10.04 | WIMPy | gehko_: You could have module unloaded it |
| 23:10.08 | citywok | just telnet in to the AMI, and log the output while making a test call. you'll see the bridge action. |
| 23:11.03 | RAFAIP | but it this doesnt work for my web application |
| 23:11.17 | gehko_ | WIMPy: haha thanks, i am such a n00b |
| 23:11.28 | citywok | RAFAIP: why not? |
| 23:11.45 | citywok | you want to find out what the other channel name is. that's a fairly simple way to do it. |
| 23:12.11 | RAFAIP | but i can make it automatically from my web application? |
| 23:12.24 | citywok | figure out how to do it by hand |
| 23:12.35 | citywok | then, make that happen programatically. |
| 23:13.06 | RAFAIP | mmm... i dont understand |
| 23:13.11 | RAFAIP | :/ |
| 23:13.16 | WIMPy | RAFAIP: We don't know what YOUR application can do, but if it's connected to AMI already it shouldn't be hard to get that implemented. |
| 23:13.33 | *** join/#asterisk jetlag (~jetlag@pool-173-61-243-204.cmdnnj.east.verizon.net) |
| 23:13.48 | RAFAIP | im feeling so noob |
| 23:14.28 | *** join/#asterisk Micc (~quassel@c-24-18-20-54.hsd1.wa.comcast.net) |
| 23:14.33 | RAFAIP | i understand what you mean, but have no idea of what i have to do |
| 23:14.50 | citywok | you have to figure out how :P http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 23:14.57 | RAFAIP | well, i have to go, thanks for the help |
| 23:15.26 | Micc | Why would asterisk voicemail try to plan an unavail.slin file when it doesn't exist but there is a .wav and .WAV and .g722? Shouldn't it play whatever one it can transcode best from? |
| 23:15.30 | RAFAIP | good bye |
| 23:16.04 | citywok | Micc, does it play, but say playing .slin on the console? |
| 23:16.37 | Micc | citywok, I think it fails on the slin and plays the vm-intro |
| 23:17.14 | Micc | all I hear is "please leave a message after the tone, when done hangup or press pound" |
| 23:17.15 | citywok | mine says playing unavail.gsm even though the phone is g729 so it's transcoding |
| 23:17.24 | Micc | but that should be played after the greeting. |
| 23:17.29 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 23:17.47 | Micc | but do you actually have a gsm file? |
| 23:17.57 | citywok | pastebin an ls -lat of your voicemail folder as well as console output of it failing |
| 23:17.58 | Micc | I would expect it should give an error that the file is not found. |
| 23:18.12 | citywok | ~pb |
| 23:18.13 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 23:24.48 | Micc | citywok: http://pastebin.ca/1995069 |
| 23:26.13 | citywok | i'm assuming you are in /var/spool/asterisk/voicemail/sbe-main right? it's not under like the default context in teh file system? |
| 23:27.20 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 23:27.39 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 23:28.41 | DrDigital | Linksys PAP2T has 2 RJ11 Ports.... can basicly run two extensions right? Does anyone make a device that I could hook like 8 old devices to? |
| 23:28.50 | candrews | my gtalk.conf: http://pastebin.ca/1995071 output in asterisk: http://pastebin.ca/1995073 When I call my gtalk number, asterisk doesn't map to the context I set in gtalk.conf. |
| 23:28.59 | candrews | Why is that? What can I do to route incoming gtalk calls? |
| 23:29.01 | DrDigital | each one having their own extension |
| 23:29.01 | citywok | DrDigital: http://www.google.com/search?sourceid=chrome&ie=UTF-8&q=8+port+fxs+gateway |
| 23:32.15 | DrDigital | citywok, would something like this be good? http://www.google.com/url?sa=t&source=productsearch&cd=5&ved=0CGoQgwgwBA&url=http%3A%2F%2Fwww.ipphone-warehouse.com%2FOpenVox-A800E1-p%2FA800E10.htm&rct=j&q=8%20port%20fxs%20gateway&ei=mGXkTISrKpLSsAP0v7hm&usg=AFQjCNEZw5CYUbKym40s4kIWtfqWCHQyZA&cad=rja basicly i need to hook up 8 data devices (fax machine, 3 atm machines and a western union terminal |
| 23:32.17 | *** join/#asterisk ccesario (~ccesario@201-42-156-44.dsl.telesp.net.br) |
| 23:33.04 | citywok | Honestly it isn't something I've ever done so I wouldn't be able to say whether or not it is good :) |
| 23:34.26 | DrDigital | ive used the linksys one personally and for clients for 1-2 devices |
| 23:34.33 | DrDigital | never had someone that needed like 8 |
| 23:34.52 | DrDigital | i just dont want to add in 4 if for a few hundred dollars more i could get 1 device |
| 23:34.53 | citywok | Yea, i've used it for 1 device. But i've also never played with asterisk+fax. i just have a POTS line at each location to avoid dealing with it. |
| 23:34.59 | DrDigital | helps keep the install cleaner |
| 23:35.29 | citywok | I'd suggest talking to the mfr to find out if their cards support data quality calls |
| 23:35.30 | DrDigital | i have no issues with faxing... i got tons of those setup |
| 23:35.52 | citywok | on PAP2's ? |
| 23:35.55 | DrDigital | i normally buy from voipsupply.com |
| 23:35.56 | DrDigital | yeah |
| 23:36.16 | DrDigital | i got a location with 3 of them just because the fax machines and credit card are all over the place |
| 23:36.37 | DrDigital | this place (a grocery store) has them all in 1 spot pretty much |
| 23:37.06 | DrDigital | they do 4 faxes a month |
| 23:37.16 | DrDigital | pay roll weekly |
| 23:37.25 | citywok | gotcha. what sip provider do you use? |
| 23:37.28 | DrDigital | and they are paying $50 a month for that line |
| 23:37.30 | DrDigital | vitelity |
| 23:41.24 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:41.25 | citywok | interesting. i have a spare pap2t, i'll have to nab it and see how well it works in my environment. i've always just avoided it because it didn't used to be that great. |
| 23:42.54 | Micc | citywok, sorry I should have added a pwd to that pastebin, but its in /var/spool/asterisk/voicemail/sbe-main/701 |
| 23:43.27 | citywok | enable debugging in logger.conf and watch the debug log while making a test call. what does it say? |
| 23:44.04 | DrDigital | citywok, just make sure the extension its using has no voicemail features |
| 23:44.18 | DrDigital | http://www.amazon.com/Grandstream-GXW-4008-port-VoIP-Gateway/dp/B001I4TQ3O im wondering how that will work |
| 23:44.35 | citywok | if it's like the other grandstream devices i have, probably fairly poorly. and then in 6 - 12 months it will die. |
| 23:44.37 | citywok | ~phones |
| 23:44.37 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 23:45.51 | Micc | DrDigital, do you use a jitter buffer on both sides for faxing/credit card machines? |
| 23:46.12 | tyman | Is the cause of this error obvious to anyone with no additional info: [Nov 17 23:43:49] NOTICE[1030]: chan_sip.c:17917 handle_response_invite: Failed to authenticate on INVITE to '"Tylers MBP Line 1" <sip:100ce039@184.106.153.227>;tag=as34ce9e90' |
| 23:46.24 | Micc | DrDigital, do the pap2t's connect directly to vitelity or through your asterisk, then relay to vitelity? Are you in the audio stream or do you reinvite to vitelity? |
| 23:47.07 | tyman | outbound call gives immediate busy and this error...producing a sip debug and sip.conf paste now for posting |
| 23:47.18 | citywok | tyman: is your device registered? |
| 23:47.40 | Micc | DrDigital, I've installed about 20 grandstreams and 50 PAP2T's and SPA2102 and about 5 SPA8000's, you want to stick with linksys/cisco. Grandstreams I end up having to replace with a linksys at some point. |
| 23:47.40 | tyman | citywok: yes, and it gets inbound calls from the pstn |
| 23:47.51 | citywok | enable sip debugging and ~pb |
| 23:47.54 | tyman | only thing that changed was my location of my phone |
| 23:48.04 | tyman | from home to office |
| 23:48.20 | Micc | DrDigital, and I've tried a lot of other adapters too, like zyxel and audiocodes mp20x, all shit. |
| 23:49.03 | Micc | citywok, any idea on that voicemail issue? I'm pretty sure it was working fine up until recently. |
| 23:49.40 | citywok | Micc: enable debugging in logger.conf and watch the debug log while making a test call. what does it say? |
| 23:51.23 | Micc | citywok, that will be a lot of data for this production box. |
| 23:51.25 | tyman | citywok: http://pastebin.com/9rNSihL8 |
| 23:52.19 | Micc | console already has debug on |
| 23:52.26 | Micc | should I turn up the verbosity? |
| 23:52.47 | citywok | you have to modify logger.conf to tell it to send debug to the console/file |
| 23:53.16 | DrDigital | honestly, im not the one who configures the stuff... I got a guy that does all that, I just buy and install it so i dont know every little aspect of the configuration, i can find out if you want to email me and when i get the answers i can forward them to you |
| 23:53.37 | DrDigital | Micc, they connect to my asterisk system |
| 23:54.35 | DrDigital | okay so linksys DOES make an 8 port |
| 23:54.45 | Micc | yes |
| 23:54.48 | DrDigital | id rather stick with linksys since the small guys have worked flawless |
| 23:54.48 | citywok | Micc you can probably grep for voicemail stuff |
| 23:55.02 | DrDigital | voipsupply doesnt ahve anything bigger then 2 port it seems |
| 23:56.05 | citywok | if you really like ordering from them you can probably ask a rep there, they can probably get it |
| 23:56.59 | Micc | ipphone-warehouse.com should have it too. |
| 23:57.15 | DrDigital | well i also just learned that i can get a lot of this stuff from bestbuy too, i havent tried yet |
| 23:57.17 | tyman | citywok: within sip debug, line 123, I find SIP/2.0 401 Unauthorized |
| 23:57.46 | Micc | they won't have the spa8000 |
| 23:57.51 | citywok | look @ line 263, proxy auth required. not sure what that's about |
| 23:58.30 | Micc | citywok, thats usually the UA just getting the digest then authenticates correctly in the next packet. |
| 23:59.14 | tyman | Micc: yes, I googled that error and read just that |
| 23:59.17 | Micc | or some other form of authentication. I don't know how all that stuff works, but I see that on most devices registering. |
| 23:59.22 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 23:59.30 | DrDigital | Micc, wanna wager? |
| 23:59.51 | Micc | DrDigital, about bestbuy having an spa8000? |
| 23:59.53 | tyman | I'm going to read the sip rfc next week...i hate being a victim... |