| 00:05.18 | igcewieling | Can anyone thing of why Asterisk might think this peer is behind NAT? the peer [igv0468] has nat=never and directmedia=yes |
| 00:05.18 | igcewieling | [Feb 12 18:56:43] Sending to xxx.xx.96.214:5060 (NAT) |
| 00:05.18 | igcewieling | [Feb 12 18:56:43] Found peer 'igv0468' for '7185132496' from xxx.xxx.96.214:5060 |
| 00:07.30 | apb1963 | igcewieling Is there any chance you have more then one entry for igv0468? |
| 00:08.12 | soFh | Hi |
| 00:08.40 | soFh | i've an issue with asterisk over IAX |
| 00:08.51 | soFh | ast-A sends call to Ast B using IAX |
| 00:09.04 | igcewieling | apb1963: that appears to be the most intelligent thing you've said all day. There is not a duplicate in sip.conf, but I was experiementing with realtime sip a month or two ago and it might be in there. |
| 00:09.24 | soFh | voice from AST-B to AST-A is perfect , while voice from Ast-A to Ast-B is broken and almost impossible to understand |
| 00:09.35 | soFh | i tried with both trunk=yes and trunk=no still same |
| 00:09.52 | soFh | while same boxes give great quality on SIP |
| 00:10.34 | soFh | can any one help or did somebody face what i am having ? |
| 00:11.17 | igcewieling | soFh: few people use IAX, why are you using IAX when SIP works for you? |
| 00:11.37 | soFh | igcewieling, only to save bandwidth |
| 00:11.57 | soFh | can't use sip , infact sip can't be used without openvpn which increases overhead a lot |
| 00:11.58 | apb1963 | igcewieling Thanks! |
| 00:12.10 | igcewieling | soFh: I wish you the best of luck, but because so few people use IAX you'll have trouble getting help. |
| 00:12.26 | igcewieling | sip can be used behind NAT and firewalls. |
| 00:12.39 | soFh | but when sip is blocked by ISP |
| 00:12.41 | soFh | then how ? |
| 00:12.47 | soFh | nothing but a vpn |
| 00:13.00 | soFh | which costs me around 42kbit/s for a single call with g729 |
| 00:13.24 | soFh | while in iax2 , i get same call on 12kbit/s |
| 00:13.43 | soFh | but this one side broken voice just came out of nowhere |
| 00:14.53 | soFh | and thanks for the wish ..i hope out of 216 people may be some body uses IAX |
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| 00:18.00 | Mon|A|rch | are there any speech recognition apps for asterisk? or all they all proprietary and hidden away |
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| 00:18.34 | Mon|A|rch | i see there's the google version |
| 00:19.05 | Mon|A|rch | but that's for TTS |
| 00:19.25 | leifmadsen | PocketSphinx has been said to work |
| 00:19.29 | leifmadsen | has not tried actively |
| 00:19.40 | Mon|A|rch | i see |
| 00:19.41 | Mon|A|rch | thank you |
| 00:20.02 | leifmadsen | speech to text is incredibly difficult to implement, so it is typically only done through commercial sponsorship |
| 00:20.41 | navaismo | Mon|A|rch, google also have for speech recognition |
| 00:20.43 | Mon|A|rch | makes sense |
| 00:21.00 | Mon|A|rch | navaismo, it's true, but that returns mp3s of your text |
| 00:21.11 | Mon|A|rch | I'm just looking to recognize a 'yes' and a 'no' |
| 00:21.15 | Mon|A|rch | i don't need the TTS features |
| 00:21.19 | Mon|A|rch | or want them for that matter |
| 00:23.00 | navaismo | uh? if you think the TTS need to be used with the speech engine nope thats not true you can use the engine separated from googles tts |
| 00:23.49 | navaismo | i was using the engine to turn on the outputs of the raspberry telling "led 1 on" or "led 2 on" |
| 00:25.25 | Mon|A|rch | i see |
| 00:25.26 | Mon|A|rch | that's neat |
| 00:25.31 | Mon|A|rch | maybe I'll give that a whirl then |
| 00:26.43 | navaismo | if you are using the zaf example the ${utterance} has the text of your speech |
| 00:29.42 | Mon|A|rch | neat |
| 00:29.48 | Mon|A|rch | how well does it appear to work? |
| 00:33.00 | navaismo | i tested in spanish and works very good, sonce ist "free", you just need to adjust the confidence |
| 00:33.17 | navaismo | lets say at 70% |
| 00:33.54 | Mon|A|rch | is .7 about what you've noticed the threshold to be? |
| 00:34.20 | Mon|A|rch | zaf claims that .9 and above is the success range |
| 00:34.32 | Mon|A|rch | or is that just the definite success range |
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| 00:42.30 | navaismo | well maybe in english language the .9 its a success, but im using spanish and i must down to .5 |
| 00:42.51 | navaismo | the word was ok but the confidence was lower tha .9 |
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| 00:51.29 | Mon|A|rch | i see |
| 00:51.37 | Mon|A|rch | well, I'll find out |
| 00:51.57 | Mon|A|rch | also, I'm glad you've tried all of the things I'm trying now |
| 00:52.01 | Mon|A|rch | very convenient |
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| 00:52.26 | igcewieling | If Adtran fixed the quality reporting on their boxes, productivity at ITSPs would increase 20% at least. |
| 00:52.39 | igcewieling | sticks a few more pins into the adtran voodoo doll. |
| 00:53.40 | S[h]O[r]T | if im configuring a DID how can i match the 1 before the number. ex 5555555555 and 15555555555. i tried like [1]555 or just X555, etc but doesnt seem to work |
| 00:54.02 | S[h]O[r]T | that way i dont have to make two inbound routes for 1 number. |
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| 00:55.29 | igcewieling | S[h]O[r]T: We usually do an exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) then do the real processing in exten => _1NXXNXXXXXX |
| 00:55.35 | leifmadsen | +1 |
| 00:55.53 | leifmadsen | I actually create all my pattern matches and then do a GoSub() to the real work |
| 00:56.09 | JerJer | i have an asterisk 1.9.19.1 process that is taking 500% and more cpu with 40 sip calls and a couple AMI clients … how should I diagnose the high cpu issue? |
| 00:56.10 | leifmadsen | exten => _1NXXNXXXXXX,1,GoSub(callcontrol,start,1(${EXTEN})) |
| 00:56.16 | leifmadsen | JerJer: 1.9 eh? nice. |
| 00:56.23 | JerJer | er 1.8 19 |
| 00:56.28 | leifmadsen | heh assumed :) |
| 00:56.29 | JerJer | damn keyboard |
| 00:58.13 | JerJer | i've tried attaching with gdb but nothing jumps out as odd |
| 00:59.13 | S[h]O[r]T | thnx |
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| 02:19.50 | blizzow | Is there a utility to check syntax of a dialplan file before putting it in place? |
| 02:21.05 | [TK]D-Fender | nope |
| 02:21.12 | [TK]D-Fender | Failure happens run-time |
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| 02:23.47 | blizzow | No wonder I find my life exciting and normal people find me boring. |
| 02:24.12 | blizzow | Thanks [TK]D-Fender |
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| 02:57.23 | ChannelZ | Wonder if there's any syntax highlighting dictionaries for text editors that could at least point out glaring errors |
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| 03:22.41 | blizzow | What am I doing wrong with my label in this GotoIf statement? http://pastebin.ca/2313299 |
| 03:22.42 | blizzow | I'm getting an error saying there is no such label 'dial303' in extension '.......' in context 'queuedial-ctd' |
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| 03:25.43 | blizzow | found it. Let Me Google That For mYself. lmgtfy.com. |
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| 03:31.49 | ChannelZ | extracomma |
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| 04:11.48 | din3sh | mrning all |
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| 04:36.29 | drjfreeze | Hi |
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| 04:42.56 | blizzow | Am I doing the labels wrong on lines 7 & 8 here?: http://pastebin.ca/2313305 My dialing plan seems to be hitting line 7 even though I can see $AREACODE is set to 303. |
| 04:45.19 | ChannelZ | line 8 is missing the close ) on your label |
| 04:45.33 | ChannelZ | (for one) |
| 04:49.21 | ChannelZ | As for your logic issue, I don't see where you are treating 303 as special, and you're always going to get to the dialPIT priority regardless (although there is something going on above what you pasted which I don't know) |
| 04:51.36 | ChannelZ | also on line 7 I don't know what your ?leavePIT is for |
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| 04:56.13 | blizzow | I think I figured it out. http://pastebin.ca/2313309 I was able to get rid of line 8 by tacking a goto on the end of line 7. It's supposed to strip the area code and check against known area codes in a region and adjust callerID to our regional office number. Failing to headquarters in the 303. |
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| 05:01.30 | blizzow | I guess I didn't. :( |
| 05:07.55 | din3sh | grrrr |
| 05:08.31 | din3sh | i continue to have HDLC errors on my 2 spans of dual E1 card |
| 05:08.55 | din3sh | changed cables |
| 05:09.04 | din3sh | re compiled dahdi |
| 05:09.11 | din3sh | updated asterisk |
| 05:15.10 | drjfreeze | Having problems with polycom phones not registering |
| 05:15.11 | drjfreeze | Registration failed User: 503, Error Code:480 Temporarily not available |
| 05:15.20 | drjfreeze | I don't see any sip traffic from the phones |
| 05:15.40 | drjfreeze | but, tcpdump sees traffic when using nc |
| 05:20.23 | drjfreeze | anyone know why a polycom phone would not be sending sip traffic to register? |
| 05:22.08 | ChannelZ | blizzow: You've either uncovered a new syntax to Set I'm not aware of with lines 8 and 16, or they are not doing what you think they are supposed to. |
| 05:23.31 | ChannelZ | But as I pointed out before, you're always going to hit lines 8 and 16 regardless if it gets as far as the part of the dialplan you are showing. |
| 05:25.20 | ChannelZ | I'm guessing you are wanting, on the n(dialBAL) priority for example, for it to set the CallerID and then Goto your qm-queuedial but that is not going to happen. You can't just shove the Goto in there. |
| 05:25.52 | ChannelZ | (again unless this is some magical addition to Set I'm unaware of, someone can set me straight..) |
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| 05:27.51 | ChannelZ | what you've really done is set the CallerID to "4102055265 <4102055265>),Goto(qm-queuedial,s,1" |
| 05:28.53 | ChannelZ | drjfreeze: is it sending the traffic to the right place? |
| 05:32.57 | blizzow | blizzow: It's most likely the latter, not the former. I looked at this: http://leifmadsen.wordpress.com/2011/08/12/set-and-goto-on-same-line/ but I'm having trouble wrapping my head around the syntax. |
| 05:33.36 | blizzow | I tried to shoehorn my stuff into the NoOp() three line format he shows first and that didn't work either. |
| 05:35.38 | ChannelZ | Well you can't just tack the Goto onto the end as you did, it doesn't work like that. |
| 05:36.03 | blizzow | I realize that now. Any suggestions on how I should do this? I just want to match $AREACODE, set CALLERID based on the match, and exit to the next context of my dial plan. If it doesn't match, keep going through them. |
| 05:36.43 | blizzow | I feel like I'm a pubic hair away from solving this. grr. |
| 05:38.34 | ChannelZ | Well there's a few ways you could go about it.. |
| 05:39.26 | ChannelZ | For something like that I typically would just write an AGI in some other language that made more sense :) |
| 05:40.08 | ChannelZ | But in the dialplan, lets see... |
| 05:40.16 | blizzow | Yeah, I wrote a little bash script as a backup solution, but I'm really interested in doing this as a dialplan solution. |
| 05:40.53 | blizzow | It would be helpful if there were a setif. |
| 05:41.34 | ChannelZ | well you can |
| 05:42.24 | ChannelZ | Set(FOO=$[${AREACODE}=240]?"its 240":"its not 240") |
| 05:43.15 | drjfreeze | ChannelZ: packet capture on the phone shows no traffic from the phone on port 5060 |
| 05:43.26 | ChannelZ | So you could do similar to what you are doing but use it to set a simple flag, like REGION=1, REGION=2, etc. depending on the area code. Then at the bottom after all the Set()s you set the callerID to whatever based on the flag. |
| 05:43.55 | ChannelZ | drjfreeze: you said tcpdump did show traffic of some sort. |
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| 05:48.18 | ChannelZ | blizzow: sorry I missed an IF function in there... Set(REGION=${IF($["${AREACODE}"="240"]?1}) |
| 05:48.40 | drjfreeze | ChannelZ: yes, we see it loading the config files (ftp and such) |
| 05:48.47 | blizzow | Thanks, I've been wondering what I was missing. |
| 05:48.50 | ChannelZ | So if AREACODE was 204 we set REGION to 1. Make up numbers for your various regions |
| 05:48.59 | drjfreeze | I've never seen a phone load config files and app.ld and not register |
| 05:49.22 | ChannelZ | Then when you're all done, do the logic of setting the CID based on the REGION, and then a single Goto to your queue dial extension thing. |
| 05:50.55 | ChannelZ | so Set(CALLERID(all)=${IF($[${REGION}=1]?5551212 <5551212>)}) |
| 05:51.14 | ChannelZ | at least I think I did that right, my eyes cross with all these braces and brackets with this font |
| 05:53.49 | ChannelZ | drjfreeze: is the proxy/server the phone is supposed to connect to specified in the provisioning files it's downloading? IE does it even know the right IP its supposed to be talking to (your asterisk box) |
| 05:56.05 | drjfreeze | ChannelZ: yes |
| 05:56.34 | drjfreeze | phones worked. we put in new router. phones don't register now. |
| 05:56.43 | drjfreeze | phone server can ping phones |
| 05:57.01 | drjfreeze | devices on phone network can ping phone server |
| 05:57.20 | ChannelZ | well if tcpdump on the same segment isn't showing any UDP coming out of the thing I'm not sure what to say |
| 05:57.59 | drjfreeze | we can watch traffice hit the phone server. even udp traffic on port 5060. but don't see the phones try to register to the server on that port. they do connect to the server for ftp config files |
| 05:58.03 | ChannelZ | perhaps a subtle type-o |
| 05:58.30 | drjfreeze | yeah, been looking for a possible typo... |
| 05:59.01 | blizzow | I just feel like there is an extra step there. I feel like gotoif could save me a step. |
| 05:59.21 | blizzow | I'm also now wondering if testing the negative might be a better solution. |
| 05:59.46 | drjfreeze | ChannelZ: care to see the log files during a phone boot? |
| 05:59.54 | ChannelZ | It's not like a normal language. You can't if/then/else in a sane way with multiple blocks of action |
| 06:00.24 | ChannelZ | drjfreeze: I don't know that they would make any sense to me not being a polycom person |
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| 06:03.12 | morfin | hello |
| 06:04.01 | morfin | how can i check if user hanged up when i am using call files to dial clients? |
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| 06:08.31 | din3sh | what does this mean? PRI Span: 1 > Message Type: ALERTING (1) |
| 06:15.57 | din3sh | how to we disable pri debug on a span? |
| 06:18.13 | drjfreeze | pri no debug |
| 06:18.23 | drjfreeze | pri no debug span 1 |
| 06:19.43 | din3sh | no such command |
| 06:19.48 | din3sh | on asterisk 1.8 |
| 06:20.38 | drjfreeze | core pri? |
| 06:22.30 | din3sh | no such command |
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| 07:21.48 | schmidts | good morning |
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| 07:23.26 | din3sh | ola senior shmidts |
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| 07:40.58 | apurvtwr | is this a good place to ask a question about asterisk-java? |
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| 07:42.08 | schmidts | apurvtwr i am not sure but i guess you can ask it and if we couldnt help you you could try it in asterisk-dev |
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| 07:46.45 | evincetech | #sangoma |
| 08:02.57 | apurvtwr | thanks schmidts . I am trying mute/unmute/end actions on a channel in confBridge. For this I am trying to capture confBridge join events from asteriskserver |
| 08:04.37 | apurvtwr | which is not something that AsteriskServerListener allows for. so I googled for a way to capture these events, but couldn't find anything convincing. |
| 08:05.22 | schmidts | apurvtwr take a look at the AMI interface (Asterisk manager interface) you should be able to see these events there |
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| 08:07.13 | kaldemar | asterisk-java is like a library for listening on AMI among other things. if asterisk-java does not support confbridge events, you'll have to add support for them yourself. |
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| 08:09.31 | apurvtwr | hm. thanks. schmidts and kaldemar : I see. I see confBridgeevents in asterisk-java but what can't find a listener for that. |
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| 08:15.12 | kaldemar | apurvtwr: try using AbstractManagerEventListener, the confbridge events don't seem to have their own listener yet. |
| 08:17.32 | apurvtwr | hm. let me try that. |
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| 11:30.45 | asteriskATmarmuD | hi guys, I'm struggling with faxing. I'm a bit confused. I want to use a fax machine in the most simple and reliable way. do I need to use t.38 pass-through? and as far as I know I have to disable fax detection. any help is highly appriciated. faxing works, but is far too unreliable :( |
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| 12:01.01 | din3sh | hello all |
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| 13:01.20 | Rico29 | hi all |
| 13:01.39 | Rico29 | I'm trying to put voicemail messages in sql database with odbc |
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| 13:02.29 | Rico29 | it works fine, but I have a warning in the CLI each time a leave a message or when I listen to it : |
| 13:02.34 | Rico29 | [Feb 13 14:02:04] WARNING[7668]: app_voicemail.c:3468 retrieve_file: SQL Get Data error! coltitle=category |
| 13:02.34 | Rico29 | [SELECT * FROM AST_voicemail_messages WHERE dir=? AND msgnum=?] |
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| 13:15.15 | apurvtwr | kaldemar: there? |
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| 13:36.58 | wdoekes | Rico29: using VM_CATEGORY ? |
| 13:37.30 | Rico29 | wdoekes> no |
| 13:37.44 | Rico29 | I don't know what VM_CATEGORY is |
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| 13:49.19 | catphish | just got a cisco 7975 and i'm so incredibly confused by where to start! |
| 13:49.27 | wdoekes | Rico29: you can try removing the category from the voicemail table, or describe table info and see what might cause asterisk to fail on it |
| 13:54.52 | *** join/#asterisk mahiti-irc (~mahiti@202.83.16.183) |
| 13:55.01 | mahiti-irc | hi |
| 13:55.16 | mahiti-irc | I am using Digium TE220 card |
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| 13:57.04 | mahiti-irc | The PRI provider has given a H/W EPABX box |
| 13:57.30 | mahiti-irc | I have connected the PRI to a Asterisk where the TE220 resides |
| 13:58.38 | mahiti-irc | Is it possible to use Asterisk to redirect imcoming calls to the H/W EPABX box |
| 13:58.47 | mahiti-irc | by using TE220?? |
| 13:59.11 | [TK]D-Fender | yes |
| 13:59.52 | mahiti-irc | how should i do it? |
| 13:59.54 | catphish | wow, cisco 79 phones can only be configured with tftp :'( |
| 14:00.44 | mahiti-irc | As of now Span 1 connected to PRI |
| 14:00.56 | mahiti-irc | So how do i redirect them to Span 2? |
| 14:01.35 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 14:02.08 | jsmith | mahiti-irc: Put span two in its own channel group (group= in chan_dahdi.conf) |
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| 14:02.27 | jsmith | mahiti-irc: Then Dial(DAHDI/g2/5551212), where 5551212 is the number you want to dial |
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| 14:03.08 | mahiti-irc | jsmith, do you mean to say 5551212 is an external phone no.? |
| 14:03.27 | jsmith | mahiti-irc: Yeah, whatever number you're trying to connect the call to |
| 14:03.35 | mahiti-irc | but no, its not like that |
| 14:03.37 | mahiti-irc | let me explain |
| 14:03.46 | jsmith | OK, I'm obviously missing something |
| 14:04.05 | jsmith | Asterisk is sitting between your telco provider and the EPABX box |
| 14:04.22 | jsmith | Span 1 is configured for PRI CPE, and is connected to the telco, right? |
| 14:04.27 | mahiti-irc | PRI ->TE220 (span1) -> Asterisk ->TE220(span2 -> H/W EPABX |
| 14:04.32 | mahiti-irc | ya right |
| 14:04.41 | jsmith | Span 2 is configured for PRI NET, and is connected to the EPABX |
| 14:04.47 | mahiti-irc | ya right |
| 14:05.12 | jsmith | OK, and I'm assuming you have something like group=1 for span 1 and group=2 for span 2? |
| 14:05.21 | mahiti-irc | no |
| 14:05.25 | mahiti-irc | do i have to? |
| 14:05.28 | jsmith | (so that you can reference an entire group, and not individual channels on the PRI) |
| 14:05.32 | jsmith | Makes it a whole lot easier |
| 14:05.44 | jsmith | By setting up channel groups, you can just say "Dial out the first available channel on this group" |
| 14:05.59 | jsmith | Otherwise, you'd have to write the logic to say "Try channel 1, then channel 2, then channel 3, etc." |
| 14:06.14 | jsmith | I *strongly* recommend putting each span into its own channel group |
| 14:06.18 | mahiti-irc | ok will do the grouping, but how to dial to epabx? |
| 14:06.31 | mahiti-irc | i mean redirect |
| 14:06.40 | jsmith | OK, so when calls come into Asterisk on Span 1, they'll go to whichever context you have set for that channel |
| 14:06.47 | mahiti-irc | ok |
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| 14:06.54 | jsmith | So let's say you have context=from_telco in chan_dahdi.conf for the span 1 channels |
| 14:07.00 | mahiti-irc | ok |
| 14:07.08 | jsmith | If you want to just pass them straight through, you'd do something like this: |
| 14:07.12 | jsmith | [from_telco] |
| 14:07.31 | jsmith | exten => _XXX.,1,Dial(DAHDI/g2/${EXTEN}) |
| 14:07.45 | mahiti-irc | oh ok |
| 14:07.58 | jsmith | That says "take any incoming call with a DNIS of four or more digits, and pass it down the first available channel in group 2" |
| 14:08.07 | jsmith | Does that make sense? |
| 14:08.30 | mahiti-irc | that will work ya |
| 14:08.38 | mahiti-irc | ya makes sense |
| 14:08.56 | jsmith | Of course, you can get a bit fancier as well |
| 14:09.07 | jsmith | Personally, I'd add a bit of logging just to make debugging easier |
| 14:09.32 | jsmith | exten => _XXX.,1,Verbose(2,Received a call with DNIS ${EXTEN} from the telco) |
| 14:09.44 | mahiti-irc | oh ok |
| 14:09.44 | jsmith | exten => _XXX.,n,Dial(DAHDI/g2/${EXTEN}) |
| 14:09.51 | mahiti-irc | great |
| 14:10.00 | jsmith | That way, you'll see it on the console when a call comes in |
| 14:10.23 | mahiti-irc | for call logs, ya that wud be great... |
| 14:10.58 | jsmith | Well, call logs are configured in cdr.conf (or cel.conf, if you want call event logs) |
| 14:11.12 | jsmith | This is just for debugging purposes as you're getting started |
| 14:11.26 | mahiti-irc | ok ok, got it.. |
| 14:11.44 | mahiti-irc | thanks for your answer, will try those |
| 14:11.57 | mahiti-irc | let me chk it and get back to you |
| 14:11.59 | jsmith | Give it a shot, and then come back if you run into problems, and we'll try to troubleshoot further |
| 14:12.03 | jsmith | :-) |
| 14:12.09 | mahiti-irc | ya thanks :) |
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| 15:08.45 | Tuju_ | hi, any idea why my SIP calls always show nnn@<ipaddress> instead of nnn@example.com ? |
| 15:11.29 | wdoekes | Tuju_: tried fromdomain= ? |
| 15:11.47 | Tuju_ | have not but just googled it and it's not set in my conf. |
| 15:14.09 | Tuju_ | ha, it worked! :) damn, that has annoyed me months. :) |
| 15:14.15 | Tuju_ | very nice. |
| 15:14.24 | AkkerKid | ...I Hate Microsoft SQL and their damn drivers that refuse to use the password i put in odbcinst. |
| 15:14.35 | James87 | tuju where did you set that? i have the same issue |
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| 15:14.57 | madhatt3r | hello *ers |
| 15:15.10 | Tuju_ | James87: all my calls always showed 555123@212.11.22.555 instead of my @tuju.fi |
| 15:15.31 | James87 | i have the same issue, but what exactly did you change to solve it? |
| 15:15.35 | Tuju_ | my friend called me and my phone used to show ip address, now it shows tuju.fi |
| 15:15.49 | Tuju_ | James87: see your sip.conf, search string fromdomain |
| 15:16.08 | Tuju_ | i set it fromdomain=tuju.fi and restarted my asterisk. |
| 15:16.11 | jmetro | is there a top-down strategy for asterisk queus? |
| 15:16.38 | Kobaz | ugh |
| 15:16.43 | Kobaz | REPLACE only does one character apparently |
| 15:16.45 | James87 | ok thnx will give that a try |
| 15:16.59 | Kobaz | is there a string replace, like i can subst one string with anohter |
| 15:17.14 | Tuju_ | actually i should have set it to iki.fi which does redireting for me. |
| 15:17.26 | bulkorok | Kobaz: CUT?! |
| 15:17.30 | bulkorok | just a guess |
| 15:17.42 | Kobaz | cut is extract |
| 15:18.05 | bulkorok | yeah... extract what you want and add what you need... |
| 15:18.07 | Kobaz | i guess you could do some convoluted dialplan using cut to accomplish a string replace |
| 15:18.20 | Kobaz | it would be easier to just write REPLACE_STRING |
| 15:18.29 | jmetro | if you are using vim |
| 15:18.36 | jmetro | unless you mean dialplan |
| 15:18.42 | Kobaz | dialplan |
| 15:18.47 | Kobaz | why would i be asking about vim |
| 15:19.11 | jmetro | well i saw "cut" and "replace" and thought it was a linux question until i scrolled up |
| 15:20.07 | bulkorok | but REPLACE should do it... |
| 15:20.11 | wdoekes | Kobaz: STRREPLACE |
| 15:21.04 | bulkorok | ah |
| 15:21.12 | wdoekes | added in ast10 |
| 15:21.26 | bulkorok | ah... |
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| 15:21.41 | wdoekes | easily backported though.. I think you can just copy paste the bits |
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| 15:22.46 | bulkorok | jmetro: check waht queues.conf comments say |
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| 15:29.12 | Rico29 | I'm looking for mysql CDR schema, but I can't find it in the source code |
| 15:29.24 | Rico29 | cans omebody tell me where I can faind it (for 1.8) ? |
| 15:29.26 | Rico29 | thanks |
| 15:32.06 | Kobaz | wdoekes: ah, i'm on 1.8 |
| 15:32.42 | catphish | can't persuade this cisco enterprise phone to register :( |
| 15:33.16 | jmetro | "Linear" looks like the Top-down queue strategy I want, but i'm wondering if persistentmembers=yes will keep them in the correct order. |
| 15:33.24 | jmetro | and how it handles members logging in/out |
| 15:33.41 | jmetro | since I always want them logging in to the same "spot" in the queue hierarchy |
| 15:34.12 | wdoekes | Rico29: http://pastebin.com/82UW0MfR |
| 15:34.27 | jmetro | Catphish: does it have SIP firmware on it |
| 15:35.08 | catphish | jmetro: lol yes |
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| 15:35.48 | Rico29 | wdoekes> where did you find it ? |
| 15:35.54 | catphish | i think it might be failing on some of the HTTP requests it expects to be able to make |
| 15:36.11 | wdoekes | in my sql daemon |
| 15:36.33 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 15:36.41 | wdoekes | I stripped the auto_increment from the table so you don't start at 600M |
| 15:37.31 | jmetro | catphish: are you auto-provisioning or manually setting the sip server/username/password on the phone? can you see the phone touching asterisk/ |
| 15:38.37 | Rico29 | wdoekes> any reason why it's not supplied by asterisk SC anymore ? |
| 15:39.46 | catphish | jmetro: automatic config, i'm trying to build an xml config |
| 15:39.53 | Rico29 | another question : what is the difference between cdr_odbc and cdr_adaptive_odbc ? |
| 15:39.55 | catphish | jmetro: i'm not seeing any attempt to contact the asterisk server |
| 15:41.24 | jmetro | If you try it manually, do you see it? |
| 15:41.48 | catphish | jmetro: i didn't even know it could be done manually :) |
| 15:42.02 | jmetro | hook the phone up, log in to its IP address and use the web interface? |
| 15:42.15 | catphish | the web interface only gives me logs |
| 15:42.30 | jmetro | no admin interface to configure paramters with? |
| 15:42.38 | catphish | nope, i don't think so |
| 15:42.50 | jmetro | what phone model? |
| 15:43.21 | catphish | 7975G |
| 15:45.01 | jmetro | can you configure the settings through the on-screen menus? |
| 15:46.19 | catphish | it appears not :( |
| 15:46.26 | catphish | so i guess i'll continue to have fun with the xmk |
| 15:46.28 | catphish | *xml |
| 15:46.37 | jmetro | the administration manual says you should be able to go through the settings and find a "Sip Configuration" menu |
| 15:46.49 | catphish | there is such a menu |
| 15:46.52 | catphish | but you can't change much |
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| 15:47.33 | jmetro | im showing a proxy and nat settings at least. |
| 15:47.57 | jmetro | then a "Line Settings" menu to setup the auth name |
| 15:48.16 | catphish | interesting |
| 15:48.19 | catphish | i'll take another look |
| 15:48.30 | jmetro | http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7975g/7_0/english/administration/guide/7975set.html#wp1190108 |
| 15:51.00 | catphish | jmetro: look at the right most column :) |
| 15:51.34 | jmetro | ah...the part where it says to use some kind of program =p |
| 15:51.41 | catphish | ya |
| 15:52.01 | catphish | in fact that program is a cisco server that would configure the phone in a corporate environment |
| 15:52.13 | catphish | i on the other hand have to write the xml manually |
| 15:52.28 | jmetro | right |
| 15:52.40 | jmetro | so basically the program probably generates that xml and just has a fancy UI |
| 15:52.49 | jmetro | and deploys it to the phone when you hit submit |
| 15:52.58 | catphish | i think it's actually a server |
| 15:53.13 | catphish | and it deploys to the phone by tftp |
| 15:53.14 | catphish | but yeah |
| 15:53.38 | jmetro | I think your only option is to troubleshoot your XML then |
| 15:54.52 | *** join/#asterisk twanny796 (twanny796@c141-180.i01-4.onvol.net) |
| 15:55.13 | twanny796 | anyone can tell me what is the echo test extension at sipgate? |
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| 16:23.08 | catphish | nothing i do seems to cause this device to even try to register :*( |
| 16:25.18 | *** join/#asterisk glaz (strke@hiro.glaciuz.com) |
| 16:27.15 | glaz | hi, i've updated * 1.4 to 1.8 and for some reason my sip devices won't register anymore, I've changed the username= field to defaultuser like the warning told me to, and sip debug gives me SIP/2.0 401 Unauthorized |
| 16:27.22 | glaz | anywhere else I should look? |
| 16:27.40 | glaz | all my phones get this error |
| 16:27.54 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:28.30 | chris_n | is there any way to have asterisk log the ip of unknown peers? |
| 16:29.26 | glaz | I've removed all my sip devices and only left one to test but here is my sip.conf config: http://pastebin.ca/2313468 |
| 16:29.56 | glaz | chris_n: authorized peers? |
| 16:30.13 | jmetro | i think its the "type=friend" |
| 16:30.21 | chris_n | no |
| 16:30.29 | glaz | jmetro: 1.8 has no type friend ? |
| 16:30.42 | jmetro | im using 11 so idk |
| 16:30.48 | chris_n | I'm looking for ips associated with such as this: "Received incoming SIP connection from unknown peer to 011972599244732" |
| 16:31.18 | chris_n | the calls get dumped, but I'd sure like to ban the ip |
| 16:31.50 | jmetro | I just got the same spammer Chris |
| 16:32.03 | chris_n | has had him for a couple of days now... |
| 16:32.08 | chris_n | and changing ips |
| 16:32.27 | jmetro | luckily since hes unauthenticated he just dumps into your dead-end |
| 16:32.52 | chris_n | right; I feed him the weasel message for kicks |
| 16:33.08 | chris_n | but it sure fills the cdr |
| 16:33.08 | jmetro | my favorite is allbusy-tt or codezone |
| 16:33.34 | jmetro | i think if your verbose / debug is set high enough it should tell you his iP |
| 16:34.30 | *** join/#asterisk jasonwert (~quassel@75-134-81-98.static.aldl.mi.charter.com) |
| 16:34.42 | jmetro | you should be able to country-ban germany though [its where his IP seems to comefrom] |
| 16:39.54 | chris_n | jmetro: I'll look at both options, thanks |
| 16:40.03 | glaz | Jesus, I got a SIP OK REGISTERED, and changed something and now back to Unauthorized |
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| 16:43.20 | glaz | <PROTECTED> |
| 16:43.29 | glaz | finally. thanks chris_n :p |
| 16:45.14 | igcewieling | chris_n: "core show function CHANNEL" you can get the IP, etc from that function, use it in your dialplan and run System(something) to ban them, where something is likely iptables |
| 16:46.45 | ghost75 | somebody has cisco phone configured with display idle parameters? |
| 16:46.49 | chris_n | igcewieling: cool, I'll try that; I was hoping to have something fail2ban could trigger on; maybe that will do it |
| 16:47.29 | igcewieling | chris_n: if a phone tries to register the rejection message in the logs shows the IP, but not if the person is trying to connect without registering |
| 16:48.26 | igcewieling | Our solution is block all connections from the internet using iptables, but that is not workable for most people. |
| 16:48.30 | chris_n | right, the later is what I'd live to catch; but maybe I'm just overzealous or some such |
| 16:49.00 | chris_n | sadly I'm one of the set of most people |
| 16:49.02 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 16:50.17 | igcewieling | chris_n: the first someone hacks your server and runs up thousands of dollars of international calls to Africa is when you'll realize you CANNOT be too paranoid. |
| 16:50.26 | jzaw | chris_n, also have a look at http://www.google.com/search?q=asterisk+fail2ban+howto |
| 16:51.10 | chris_n | igcewieling: my feeling exactly |
| 16:51.17 | glaz | chris_n: secret is now remotesecret |
| 16:51.44 | chris_n | jzaw: thanks; I have fail2ban working, but am hoping to adapt it to this case |
| 16:51.53 | jzaw | :) cool |
| 16:52.16 | igcewieling | chris_n: we have multiple levels of security. 32 char random secrets, iptables, fail2ban, fraud detection in our main switch, and our carriers' fraud detection. |
| 16:53.20 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:55.49 | chris_n | yeah, once this begin happening we went back and strengthened out secrets |
| 16:57.07 | igcewieling | chris_n: other attack vectors include any Asterisk GUI (like FreePBX) and the FTP provisioning user for your phones |
| 16:57.22 | catphish | what causes "FAX session '8' failed to create timing source."? |
| 16:57.37 | chris_n | will add those to the check list |
| 16:57.59 | catphish | i suspect my dummy timing isn't working properly :( |
| 17:00.17 | ChannelZ | What fax subsystem are you using? |
| 17:01.09 | *** join/#asterisk rue_work (~rue_mohr@24-207-105-65.eastlink.ca) |
| 17:01.40 | catphish | spandsp |
| 17:02.36 | rue_work | answer; playtones(dial); settimeout(waitexten(10)); waitexten(10) <-- how do I modify so the dialtone stops when you press the first digit of the extension? |
| 17:04.55 | rue_work | timeout is on digit, sorry |
| 17:05.20 | rue_work | set(timeout(digit)=10) |
| 17:06.12 | rue_work | dont know of anywhere I can put a stopplaytones |
| 17:10.20 | [TK]D-Fender | rue_work, Record it as a file and background it |
| 17:10.52 | [TK]D-Fender | And who wants 10 seconds between digits? |
| 17:11.57 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 17:14.43 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 17:20.57 | igcewieling | [TK]D-Fender: Old people. |
| 17:21.37 | Changos | Hi guys, I have a really big problem right now for me. I try to configure E1 Card, the installation is succesfull, Asterisk recognizes the E1 card, but I can get outbound calls |
| 17:21.57 | Changos | http://pastebin.ca/2313496 |
| 17:22.48 | Changos | I suspect that the key is in the file extensions.conf, but I find on many page on Internet and a didn't find some clear |
| 17:22.58 | WIMPy | And you don;t want them to work? :-) |
| 17:23.02 | [TK]D-Fender | Changos, Your [e1] context has nothing usable in it |
| 17:23.24 | [TK]D-Fender | Changos, It is broken. Go fix it |
| 17:25.23 | *** join/#asterisk bipolar (~bipolar@204.186.46.94) |
| 17:26.25 | Changos | [TK]D-Fender: I can find some clear form to configure it correctly. |
| 17:26.33 | Changos | :s |
| 17:26.45 | rue_work | [TK]D-Fender, we have slow dialers over here |
| 17:27.06 | kaldemar | Changos: what is your problem exactly? dialing out with DAHDI? |
| 17:27.16 | rue_work | so with a backgrounded file, play will stop as soon as a digiti is pressed? |
| 17:27.31 | Changos | kaldemar: exactly |
| 17:27.40 | kaldemar | Changos: the notice says that you don't have a matching extension in [users]. |
| 17:29.40 | kaldemar | Changos: you don't even have an extension that would dial out using DAHDI in your whole dialplan. |
| 17:29.49 | [TK]D-Fender | Changos, Lok at your dialplan. THere is nothign there to match. Go fix it. |
| 17:30.06 | igcewieling | rue_work: you can also look at DISA |
| 17:30.29 | Changos | kaldemar: how can I add this lines on dialplan ? |
| 17:30.51 | WIMPy | Changos: Use an editor. |
| 17:31.03 | WIMPy | And take a look at dialplan 101 in |
| 17:31.08 | WIMPy | ~book |
| 17:31.08 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:32.01 | Changos | WIMPy: I know, I know very good how work GNU/Linux |
| 17:32.10 | WIMPy | Err. Why is that 3rd edition? Wasn't the 4th released officially? |
| 17:32.36 | WIMPy | Changos: Who is talking baout GNU/Linux? |
| 17:32.56 | Qwell | WIMPy: "released" |
| 17:33.05 | Qwell | It's merely publicly available. |
| 17:33.20 | [TK]D-Fender | Changos, extensions.conf <------- |
| 17:33.25 | Changos | [TK]D-Fender: the problem is that I can't find some howto about configure correctly my extensions.conf. |
| 17:33.32 | [TK]D-Fender | Changos, Go read the book. This is your dialplan go learn to set it up |
| 17:33.35 | [TK]D-Fender | ~book |
| 17:33.36 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:33.37 | [TK]D-Fender | ^^^ |
| 17:33.42 | Changos | ok |
| 17:33.44 | Changos | a read it |
| 17:33.56 | Changos | I* |
| 17:34.12 | igcewieling | Changos: telephones are difficult, don't expect asterisk to be easy. |
| 17:34.15 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 17:36.13 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 17:37.15 | rue_work | disa, ok |
| 17:37.42 | Changos | igcewieling: I'm relax, I'm reading, I'm not noob about GNU/Linux, I know about Asterisk, but always work with FreePBX, and this is my first time that I work without it. But I see problem with correct documentation about files config, examples, etc. just right know read the dfinitive guide Asterisk |
| 17:38.42 | igcewieling | Changos: Using FreePBX is like programming in BASIC -- it ruins you for life. |
| 17:38.51 | catphish | woo, got the 7975 to *try* to register |
| 17:39.00 | WIMPy | likes BASIC |
| 17:39.03 | catphish | but it doesn't seem to respect 401 www-authenticate |
| 17:43.50 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 17:48.53 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 17:55.36 | *** join/#asterisk Hrnec (~Miranda@62.197.198.38) |
| 17:57.12 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 17:57.37 | nunne | anyone have experience with phones trying to connect to an asterisk 11 behind a freebsd ipfilter firewall? |
| 17:58.31 | nunne | I can't get the phones to register properly.. setting different source ports on the phones manually works for a few minutes.. but the firewall should translate this more "gracefully" i think :/ (i have other phones on the same box behind nat - no problem with them). |
| 17:59.01 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:59.27 | igcewieling | nunne: leave all the phone settings the default, port forward 5060 on your firewall to your internal asterisk IP. Do NOT use any SIP specific settings in your firewall |
| 17:59.59 | nunne | igcewieling: i have no control over the firewall.. but the company that has has assured me no SIP ALG etc is running |
| 18:00.19 | nunne | igcewieling: the asterisk is on a public IP.. its the phones in question that are behind nat |
| 18:00.40 | nunne | and i'm only having issues with the phones behind the freebsd box with ipfilter |
| 18:00.53 | igcewieling | nunne: assuming you have Asterisk configured correctly they should "just work" |
| 18:01.13 | nunne | real weird issues like registration without port, but still showing up as reachable.. |
| 18:01.14 | ChannelZ | You might have to turn on qualify and set the timer fairly low, THEIR firewall might be clearing out the NAT table quickly |
| 18:01.42 | igcewieling | nat=yes qualify=yes no bindaddr, no bindport, no localnet= and no externip= |
| 18:02.01 | nunne | qualify is on. i can test lowering the value.. but it's not a problem of that.. if they showed up as unreachable i would be "happy".. but the registration never even comes in |
| 18:02.14 | nunne | and on the phones it's either status 408 or 503.. a nice mixture :P |
| 18:02.40 | ChannelZ | well if you're not even getting registrations, there's not much you can do not being in control of the remote network/device |
| 18:02.48 | nunne | igcewieling: i have bindaddr.. but thats because it has 2 NICS/IPS (the box i mean). |
| 18:03.09 | igcewieling | nunne: having 2 nics is NO reason whatsoever for using bindarrt. |
| 18:03.39 | nunne | ChannelZ: I don't think there is anything wrong with my box.. I'm just wonder if anyone has experience with freebsd ipfilter that could give any input.. maybe it's a common "problem" with that setup :/ |
| 18:03.39 | igcewieling | bindaddr is almost never useful, in the few cases it is useful are almost always when you have two IPs on the same subnet on the same nic. |
| 18:04.24 | *** part/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 18:04.45 | nunne | I can try commenting that one out. but as I said I have other phones behind other firewalls that work flawlessly. |
| 18:05.41 | igcewieling | nunne: even if bindaddr isn't causing THIS issue, it will cause other issues |
| 18:06.16 | nunne | igcewieling: removing it ;) |
| 18:07.50 | igcewieling | nunne: if the registration packets are not getting to asterisk then the problem is NOT with asterisk. |
| 18:07.56 | Hrnec | Hi, is it possible to make a conference call from web through Asterisk? I mean like 2 users from Chrome calling into a conference room. I have tried similar with FreeSWITCH but failed. Have you guys tried this? Does it work? Thanks a lot for answers! |
| 18:08.34 | *** join/#asterisk lorsungcu_ (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 18:08.39 | nunne | igcewieling: I will make a tcpdump just to be sure.. because right now I'm trusting sip debug on |
| 18:10.18 | igcewieling | Hrnec: Asterisk does not have a GUI, therefore your question doesn't make much sense. |
| 18:11.00 | mjordan | I think he's referring to WebRTC. |
| 18:11.12 | Hrnec | mjordan: Exactly, sorry I haven't mention it. |
| 18:11.14 | mjordan | Hrnec: yes, in Asterisk 11, it's possible. |
| 18:11.36 | igcewieling | mjordan: sounds like he wants something which "just magically works". |
| 18:11.44 | mjordan | then prepared to be disapointed! |
| 18:11.45 | mjordan | :-) |
| 18:11.54 | mjordan | Chrome is still a moving target. |
| 18:12.10 | mjordan | I'm hopeful things will reach a more quiescent state in April |
| 18:12.49 | mjordan | Hrnec: https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support |
| 18:13.30 | Hrnec | igcewieling: I am referring to WebRTC and don't want anything that "just magically works". I know SIP quite a lot, but I haven't worked with Asterisk since 1.8 version, because I am developing things under FreeSWITCH mainly. But now I have come to this, I need at least SOME support for WebRTP and FS don't support SRTP so... |
| 18:14.05 | navaismo | Hrnec, you can install the doubango webrtc2sip media gateway to use any sip legacy pbx |
| 18:14.19 | igcewieling | Hrnec: sorry. Your question was worded like a total n00b. 8-| |
| 18:15.01 | navaismo | Hrnec, i only tested with asterisk and for audio so far is good |
| 18:17.16 | *** part/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 18:17.20 | Hrnec | igceqieling: *WebRTC. Yeah it's allright, I read it afterwards and it looks like ... well your reaction is quite appropriate. |
| 18:17.37 | Hrnec | igcewieling: *WebRTC. Yeah it's allright, I read it afterwards and it looks like ... well your reaction is quite appropriate. |
| 18:17.37 | Hrnec | d |
| 18:18.09 | Hrnec | igcewieling: Sorry I'm in quite a rush. |
| 18:18.39 | Hrnec | mjordan: Thanks for help. That's what i wanted to know. In FS - even audio doesn't work, because of SRTP. |
| 18:20.58 | pigpen | I have never done this, but I figured why not post it here: Could I use a Digium PRI card to provide channels to another * box with it's own PRI card? so something like: Telco (PRI Service) <--->Digium--Astiersk---Digium<--->Digium---Asterisk |
| 18:21.26 | pigpen | Sure, I could use sip or an iax trunk, but this way, I don't have any fax issues. |
| 18:21.39 | WIMPy | pigpen: Sure. |
| 18:21.51 | pigpen | k. that is what I thought, but have never done it. |
| 18:22.00 | pigpen | long time to see bty. |
| 18:22.27 | pigpen | and thanks |
| 18:23.29 | din3sh | i continue to experience HDLC errors on my 2 spans, E1---->asterisk----->nortel |
| 18:23.55 | din3sh | have changed cables, recompiled dahdi, recompiled asterisk |
| 18:31.42 | igcewieling | pigpen: yes. there is some DAX functionality in DAHDI, which does a "virtual patch" of clear channels between dahdi ports, this is done in DAHDI and not through Asterisk. |
| 18:32.36 | igcewieling | pigpen: Last time I did this was about 10 years ago, so I don't remember a lot about it, but check the dahdi sample config files. |
| 18:32.42 | pigpen | ah, cool. thanks. |
| 18:33.13 | pigpen | sip and iax are great and all, but hard connections like PRI's are dam handy |
| 18:33.29 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 18:33.46 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:34.15 | igcewieling | pigpen: we used it to DAX some of the CAS T-1 channels from the carrier to an Adtran, but have the rest handled by asterisk |
| 18:34.30 | pigpen | ah, yeah |
| 18:35.01 | igcewieling | we had something like this from the carrier. channels 1-15 CAS T-1, channels 16-23 B-channels, chan 24 D-chan (we had a REALLY cool carrrier) |
| 18:38.42 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:38.42 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:45.29 | TSM | is there a setting to make my sip trunk pass back the received Remote-Party-ID it gets |
| 18:50.46 | igcewieling | trustrpid=yes |
| 18:50.46 | igcewieling | sendrpid=pai |
| 18:50.46 | igcewieling | rpid_update=yes |
| 18:50.51 | igcewieling | try reading sip.conf.sample |
| 18:51.42 | igcewieling | to use rpid instead of paid set sendrpid-yes instead of pai |
| 18:51.50 | glaz | is there a way for asterisk to tell what RTP ip the remote peer should use? |
| 18:52.29 | igcewieling | glaz: it is in the SDP info of (usually) the INVITE |
| 18:53.09 | TSM | igcewieling: ahh i was reading the page on voip-info which is prob out of date |
| 18:53.12 | glaz | I know, and one of our clients sends us the wront RTP server to use, so they hear us but we can't hear them |
| 18:53.27 | glaz | I'm just trying to help the client to send us the right FTP info |
| 18:53.29 | glaz | RTP* |
| 18:53.57 | glaz | he has two NICs and he sends us the ip of the wrong NIC |
| 18:53.58 | igcewieling | TSM: ALWAYS verify info on voip-info with secondary sources like the Asterisk book the Asteisk CLI show appliction / show function, or the sample config files |
| 18:54.20 | igcewieling | glaz: is he using Asterisk? |
| 18:55.17 | glaz | igcewieling: kinda, he's using elastix which is asterisk based |
| 18:55.27 | TSM | that worked like a dead donkey, mmmm, annoying |
| 18:55.45 | igcewieling | glaz: your problem is the classic symptom of using bindaddr in asterisk |
| 18:56.00 | igcewieling | TSM: you need 1.8 or later for this to work at all |
| 18:56.12 | glaz | igcewieling: that's what I think as well, he has bindaddr set to 0.0.0.0 though |
| 18:56.33 | glaz | and he has another SIP provider connected to his 1st NIC |
| 18:56.38 | glaz | we're on the second NIC |
| 18:56.55 | glaz | RTP works fine for his other SIP provider, but not with us |
| 18:57.16 | igcewieling | glaz: Sounds like he has bindaddt=theipofthenicforhispotherprovider |
| 18:57.33 | glaz | you think that if he has 0.0.0.0 it'd work fine? |
| 18:57.41 | igcewieling | without bindaddr the OS will pick the source addr and asterisk should figure it out. |
| 18:57.44 | TSM | igcewieling: yup on both, i can get it to work localy, i can see the RPID being received by the other server but just does not get transferred to the phone |
| 18:58.03 | glaz | iirc he had 0.0.0.0 |
| 18:58.06 | igcewieling | glaz: bindaddr=0.0.0.0 should be the same as not having it. I |
| 18:58.12 | glaz | ok |
| 18:58.27 | igcewieling | sounds like he has an OS routing problem and the OS is routing packets to you out the wrong NIC |
| 18:58.40 | glaz | ok that rings me a bell |
| 18:58.48 | glaz | cause he has one route for our switch |
| 18:59.15 | glaz | anyway, he'll figure it out I guess, they signed with us, I'm just trying to be nice |
| 18:59.55 | glaz | I'll go back to MY problem heh |
| 19:00.34 | glaz | I have this error retrans_pkt: Retransmission timeout reached on transmission, on a SIP phone on the same subnet as the asterisk box |
| 19:07.40 | *** join/#asterisk apurvtwr (7ab1f3f8@gateway/web/freenode/ip.122.177.243.248) |
| 19:08.06 | *** join/#asterisk StaRetji (~LittleAll@91.142.129.1) |
| 19:08.11 | StaRetji | Howdy |
| 19:08.24 | StaRetji | folks, is : service asterisk stop |
| 19:08.31 | StaRetji | stopping asterisk gracefully or no? |
| 19:08.59 | StaRetji | if not, how to stop asterisk gracefully via bash script |
| 19:09.36 | StaRetji | I need to stop it every day at give time, I made cronjob asterisk -rx 'core stop gracefully' |
| 19:09.40 | ChannelZ | you could asterisk -x "core stop gracefully" |
| 19:09.53 | StaRetji | thx ChannelZ |
| 19:10.05 | StaRetji | I did that, but it works if I run the script manually |
| 19:10.09 | glaz | found my problem, nat=no wasn't there |
| 19:10.15 | igcewieling | why not use "core restart gracefully" |
| 19:10.17 | glaz | what is nat= by default ? |
| 19:10.19 | ChannelZ | as for the service script I guess it depends on what it's doing. I'm sure it's sending a signal that * probably exits cleaning for |
| 19:10.27 | igcewieling | glaz: depends on the version of asterisk |
| 19:10.34 | StaRetji | for some reason, it doesn't work if i run from cronjob |
| 19:10.34 | glaz | 1.8.10 |
| 19:10.39 | ChannelZ | cleanly, not cleaning. derr |
| 19:10.41 | StaRetji | igcewieling: I need to stop it mate |
| 19:10.57 | ChannelZ | Are you giving the full path to asterisk in the cron job? |
| 19:11.04 | StaRetji | no |
| 19:11.07 | StaRetji | good point |
| 19:11.10 | igcewieling | glaz: IIRC nat defaults to yes in 1.8, but you should check the UPGRADE*.txt files (NOT just for 1.8) to be sure |
| 19:11.14 | StaRetji | like /usr/sbin/asterisk |
| 19:11.15 | ChannelZ | the environment of cron probably can't find it |
| 19:11.16 | StaRetji | ? |
| 19:11.19 | ChannelZ | yes |
| 19:11.24 | StaRetji | true, thx mate, will check it |
| 19:11.34 | Qwell | ~upgrade asterisk |
| 19:11.34 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 19:11.39 | Qwell | glaz: ^ |
| 19:12.49 | glaz | damn ubuntu runs 1.8.10 |
| 19:13.14 | jmetro | ubuntu runs 11... |
| 19:13.17 | glaz | looks like a modified version of asterisk, Asterisk 1.8.10.1~dfsg-1ubuntu1 |
| 19:13.34 | jmetro | format with debian |
| 19:13.42 | ChannelZ | My ubuntu goes up to 12. |
| 19:13.55 | glaz | I did apt-get install aterisk and I got 1.8.10 |
| 19:13.59 | ChannelZ | Anyway, build from source. |
| 19:13.59 | *** join/#asterisk nightrid3r (~kvirc@62.205.64.94) |
| 19:14.01 | glaz | on 12.04 |
| 19:14.07 | jmetro | only cause youre running the spinal-tap edition channelz |
| 19:14.26 | ChannelZ | Spinal-tap+1 edition |
| 19:17.08 | TSM | i am still having no hope getting the received RPID via a sip trunk to flow to the SIP phone |
| 19:18.27 | igcewieling | TSM: sounds like it is time to slog through some sip debug data |
| 19:19.08 | TSM | i have already |
| 19:19.24 | TSM | i can see it on the inbound from the other PBX but just does not go though to the phone |
| 19:23.01 | StaRetji | ChannelZ: mate, /usr/sbin/asterisk -r "core stop gracefully" |
| 19:23.29 | StaRetji | and asterisk still running, no connections, if I run it from console, it works |
| 19:23.42 | StaRetji | I can't see what am I doing wrong |
| 19:24.19 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 19:25.09 | *** join/#asterisk ghost75 (~ghost75@dslb-178-002-152-097.pools.arcor-ip.net) |
| 19:26.22 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 19:26.27 | TSM | douh |
| 19:26.43 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 19:27.04 | TSM | so yes ive run SIP debug and i could see the RPID being sent from the other server but nothing is being transmitted to my phone |
| 19:27.26 | ChannelZ | StaRetji: you need -rx |
| 19:28.42 | StaRetji | yep, sorry for that |
| 19:28.44 | StaRetji | works ;) |
| 19:28.46 | ChannelZ | or just -x I think should work too |
| 19:28.46 | StaRetji | thxmate |
| 19:29.21 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-177-46.red.bezeqint.net) |
| 19:35.46 | apurvtwr | I am using confbridge for conference calls using asterisk but i want to add mute/unmute functionality in it. I tried that using asterisk-java but realised that there isn't a listener for conf bridge events in asterisk-java. Tried extending AbstractManagerEventListener to listen to confBridge but couldn't catch the events there either. |
| 19:37.39 | apurvtwr | has someone here done this before? if so, how? |
| 19:38.13 | jmetro | press the mute button on the phone? do you mean mute others as the admin? |
| 19:38.37 | apurvtwr | yes mute as admin |
| 19:39.22 | *** join/#asterisk b0ot (~DynamicFa@147.177.61.44) |
| 19:41.14 | *** join/#asterisk serafie1 (~erin@76.73.167.231) |
| 19:42.14 | apurvtwr | the only events that i catch on join/hangup in confbridge are bridge events. |
| 19:45.52 | saint_ | can someone pastebin an example of PAGE() with Digium phones ? |
| 19:46.15 | Qwell | Why does the type of phone matter? |
| 19:51.26 | mjordan | apurvtwr: do you want the admin to be able to mute, or do you want an AMI connection to mute a participant? |
| 19:52.14 | mjordan | because you should be able to do both. :-) |
| 19:52.38 | apurvtwr | i want the AMI connection to be able to mute a participant |
| 19:53.22 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_ConfbridgeMute |
| 19:53.45 | apurvtwr | how do I do that? and is it possible through asterisk-java? |
| 19:54.03 | mjordan | no idea about asterisk-java. But it's certainly possible to do it through AMI. |
| 19:54.29 | apurvtwr | the way I was trying it was to capture join events of confBridge and get channels from there |
| 19:54.43 | mjordan | that's fine. You would need the channels in the ConfBridge. |
| 19:54.53 | apurvtwr | hm. i see |
| 19:55.06 | mjordan | ConfBridge has its own events for it however - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_ConfbridgeJoin |
| 19:56.45 | apurvtwr | yeh, I saw this one. I wanted to be able to perform the same using an application we have written in Java. |
| 19:57.01 | apurvtwr | anyway. thanks mjordan :-) |
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| 20:03.57 | malcolmd | i missed a bit..someone's having trouble with PAGE and Digium phones? |
| 20:04.59 | glaz | does moh from a stream has to be pcm in asterisk 1.8 ? |
| 20:10.02 | eirirs | are there huge improvements for asterisk from 1.6 to 1.8 ? and how about 11 ? |
| 20:10.29 | igcewieling | eirirs: have you checked the UPGRADE*.txt files? |
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| 20:24.02 | mjordan | https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 |
| 20:24.06 | mjordan | https://wiki.asterisk.org/wiki/display/AST/New+in+10 |
| 20:24.12 | mjordan | https://wiki.asterisk.org/wiki/display/AST/New+in+11 |
| 20:24.17 | mjordan | eirirs: ^^^ |
| 20:24.19 | pabelanger | SPAM |
| 20:24.30 | mjordan | pabelanger: is delicious |
| 20:24.35 | eirirs | o_O |
| 20:24.37 | pabelanger | om nom nom |
| 20:24.38 | eirirs | thanks sir |
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| 20:26.08 | slicknick5181 | asterisk 1.8 I can not get in-call DTMF features to work-- I have a 2 line SIP ATA and I want to be able to xfer an incoming call via DTMF to the parking lot |
| 20:26.34 | slicknick5181 | I do not have a hook/flash setting on my ata device |
| 20:30.18 | igcewieling | mjordan, being an Old One cannot send spam. |
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| 20:36.15 | jmetro | cthulu sent spam ... |
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| 20:37.31 | ChannelZ | Interesting. Just got a call from a CNAM of "477047W8VU89700" |
| 20:37.47 | jsmith | ChannelZ: Slick :-) |
| 20:38.06 | ChannelZ | Creative choice. |
| 20:38.40 | ChannelZ | And ironically it wasn't spam, it was a legitimate call (the CID number was fine) |
| 20:40.22 | ChannelZ | slidesinger: With an ATA it gets a little funkier |
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| 20:41.08 | ChannelZ | The ATA can read the DTMF and turn it into rfc for instance, or perhaps it is just passing through DTMF and your Asterisk isn't aware it's supposed to be listening |
| 20:42.40 | ChannelZ | Also you have to make sure the Dial() that initiated the bridge has the k and/or t flags for instance |
| 20:43.00 | ChannelZ | (or K/T as the case may be, see 'core show application Dial' for all the dirt) |
| 20:49.05 | AkkerKid | hey all do i have to do anything special in regards to reloading the dialplan when i make additions to func_odbc? |
| 20:49.55 | jsmith | AkkerKid: Nope |
| 20:49.59 | AkkerKid | "Reload" on the CLI doesn't seem to be picking up the changes. I can't see what else may be causing an issue |
| 20:50.05 | jsmith | AkkerKid: odbc reload |
| 20:50.31 | AkkerKid | @jsmith: on the CLI? |
| 20:50.43 | jsmith | AkkerKid: Yes, on the Asterisk CLI |
| 20:51.25 | AkkerKid | hmmm, i'm still getting no result from the referenced mysql query |
| 20:51.27 | jsmith | AkkerKid: or "module reload func_odbc.so" if you've changed func_odbc.conf, or "module reload res_odbc.so" if you're changed res_odbc.conf |
| 20:52.27 | AkkerKid | no change :( |
| 20:52.34 | AkkerKid | i must have a problem somewhere else then. |
| 20:53.22 | AkkerKid | on another topic... how do I create a variable that follows a call through sifferent contexts? |
| 20:53.28 | AkkerKid | different* |
| 20:53.39 | jmetro | i believe you have to pass it to each one when you move between |
| 20:53.49 | AkkerKid | eek! |
| 20:54.22 | AkkerKid | I guess i'll just keep doing that then... |
| 20:55.25 | ChannelZ | aren't channel variables channel variables? |
| 20:57.20 | AkkerKid | If I move a call to a Goto or GoSub, I have to pass it variables that will be collected by ${ARGn} on the receiving side... |
| 20:57.38 | AkkerKid | but the variables I set in the previous context are lost |
| 20:59.30 | jmetro | that would be awful if you couldnt pass variables that you set yourself between contexts like normal programming can. |
| 20:59.33 | AkkerKid | aha! Looks like I didn't encapsulate my SQL variables in quotes |
| 20:59.36 | igcewieling | AkkerKid: no you do not. channel variables follow the call through all contexts and extensions as long as the channel does not change |
| 21:00.09 | AkkerKid | igcewieling: I will re-test. |
| 21:07.07 | igcewieling | OOPS! "Warning: Any calls that may have been active have been dropped!" |
| 21:08.30 | AkkerKid | ok now I know I have to put non-number variables in sql queries into quotes in func-odbc |
| 21:12.03 | glaz | any reason why musiconhold doesn't want to spawn my mplayer process? |
| 21:12.09 | glaz | It was working before I upgraded |
| 21:12.18 | glaz | can't find the reason |
| 21:12.30 | igcewieling | glaz: upgrade from what to what? |
| 21:12.45 | glaz | from 1.6 to 11-latest |
| 21:12.47 | glaz | 11.2.1 |
| 21:13.01 | glaz | do you want my application= line ? |
| 21:13.01 | igcewieling | glaz: nothing in the UPGRADE*.txt files which might provide insight? |
| 21:13.12 | glaz | I coulnd't find anything on moh |
| 21:13.44 | igcewieling | only crazy people do streaming MoH, so that is about all I could suggest. |
| 21:14.11 | glaz | :) |
| 21:14.57 | glaz | /usr/bin/mplayer http://5283.live.streamtheworld.com:3690/CHMPFMAAC_SC -cache 64 -af resample=8000:0:1,channels=1 -ao pcm:file=/dev/stdout -quiet -really-quiet |
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