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| 00:38.34 | jdoe | 1.8.4 has been in rc2 for about a month now... does 1.8.3.2 include the fixes that were in it, or is 1.8.4 still an rc? |
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| 01:23.18 | seraphie | jdoe: 1.8.3.2 is only a security release. It is 1.8.3 with security fixes. |
| 01:27.15 | jdoe | that's what I figured, a month just seemed like a long time for an rc. |
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| 01:44.30 | mbowie | Howdy folks... possibly a noob question, but I have a new Asterisk 1.8.2.3 setup with a SIP trunk through Broadvox and am having a struggle with sending a callerId number via the trunk. They accept the "From:" header for this purpose, but my config is sending "Unknown". |
| 01:46.42 | mbowie | If I set "fromuser=" for the trunk, the value is passed along, but I need to be able to set it on a per-extension basis. |
| 01:47.04 | mbowie | (And setting "fromuser=" for a user doesn't seem to generate the same mojo.) |
| 01:48.14 | Kobaz | sendrpid=yes |
| 01:49.56 | mbowie | Hi Kobaz. I've added that, which sets the Remote-Party-ID header, but apparently they don't accept that. |
| 01:51.08 | mbowie | The From: header appear correct in the packets between the phone and the Asterisk server, just not from the server to the trunk provider. |
| 01:52.28 | Kobaz | if you want callerid to be in From: then turn off sendrpid and clear out fromuser |
| 01:52.40 | Kobaz | fromuser will force User to be a specific value |
| 01:52.47 | Kobaz | er... force the From: |
| 01:53.04 | mbowie | Ok... I've removed fromuser... let me see if removing sendrpid helps. Thanks. |
| 01:53.58 | mbowie | Rats... no change. |
| 01:54.04 | Kobaz | and then in dialplan Set(CALLERID(num)=xxxx); |
| 01:54.09 | Kobaz | what's sip debug show? |
| 01:54.51 | mbowie | From: "Unknown" <sip:Unknown@my.ip.address>;tag=as299bc311 |
| 01:55.04 | mbowie | Is the first packet sent to the SIP trunk. |
| 01:55.17 | Kobaz | are you setting CALLERID |
| 01:58.17 | mbowie | Hrm... I think I just fell on my sword. Hold the phone. |
| 01:59.43 | mbowie | Nope... I was fine. Let me see about adding that to my dialplan. |
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| 02:10.30 | mbowie | I fear I'm missing the bus somewhere here... perhaps I've been staring at it a bit long. (Everything else is Just Working(tm)) |
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| 02:49.30 | itsbroken | Hello, I'm upgrading my asterisk installation and I wanted to know how much I can prepare before the big switch. I know that I need to recompile libpri and dahdi, then recompile asterisk. But, does that mean I can go ahead and do the libpri and dahdi part now and it will have no effect on the running instance of asterisk? |
| 02:53.56 | Kobaz | you can recompile libpri and dahdi without affecting the running asterisk, but the next time you load asterisk, it will use the new stuff |
| 02:53.59 | Kobaz | but |
| 02:54.06 | Kobaz | the best thing to do is play with everything on a test server |
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| 03:05.26 | IridiumScaffold | Can someone confirm if one of the default Grandstream custom ringtones is a women's voice saying "You have a call waiting" |
| 03:05.31 | IridiumScaffold | on a GXP-2000 |
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| 03:51.51 | f0ner00t | Hello |
| 03:52.42 | nix8n82 | hello |
| 03:53.21 | f0ner00t | How are you doing toight nix8n82. |
| 03:53.40 | nix8n82 | I am doing fine, how about you f0ner00t |
| 03:53.43 | nix8n82 | ? |
| 03:54.41 | f0ner00t | I am alright just trying to finish up my config on webmeetme 4 |
| 03:56.11 | nix8n82 | cool, haven't touched it |
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| 03:57.04 | f0ner00t | Ahh its pretty neat but i need to get asterisk to call the module |
| 03:59.48 | nix8n82 | cool, and you are having problems loading the module? |
| 04:00.36 | itsbroken | After I do my upgrade of libpri, dahdi and asterisk is there anything I need to do besides stop asterisk, install, start asterisk? Anything else I should watch out for? |
| 04:01.19 | nix8n82 | reload the dahdi drivers? |
| 04:01.30 | nix8n82 | before you start asterisk |
| 04:01.44 | itsbroken | what do you mean? |
| 04:02.50 | nix8n82 | after you install the dahdi drivers you should have a file in /etc/init.d/dahdi |
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| 04:03.22 | f0ner00t | nix8n82: No the meetme module works I just need to setup asterisk / freepbx to access the config i installed. I already tried the webmeetme module which did not work |
| 04:03.23 | nix8n82 | from the command line you should be able to enter # /etc/init.d/dahdi stop |
| 04:04.01 | nix8n82 | then # /etc/init.d/dahdi start |
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| 04:04.11 | nix8n82 | and you should she your drivers reload |
| 04:04.16 | f0ner00t | nix8n82: I already got meetme and dadhi_dummy. |
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| 04:04.37 | nix8n82 | s/she/see |
| 04:04.43 | f0ner00t | Its installing webmeetme they have instructions on how to install the interface but not how to get asterisk / freepbx to access it. |
| 04:05.13 | nix8n82 | right I was talking to itsbroken..sorry for not being clear |
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| 04:05.26 | f0ner00t | Ahh that makes sense. |
| 04:05.28 | f0ner00t | :) |
| 04:06.32 | nix8n82 | I'm not sure what your problem is. |
| 04:07.13 | nix8n82 | I haven't tried to configure it with freepbx, or old school dialplan programming |
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| 04:12.30 | itsbroken | Thanks nix forgot about that |
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| 04:14.40 | nix8n82 | you're welcome |
| 04:18.28 | f0ner00t | goodnight. |
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| 07:02.23 | henk | hi, i'm a bit confused about the state of faxing in asterisk 1.6. i run the debian squeeze variant and all the info on the voip-info.org wiki about different methods for faxing is a bit overwhelming... my sip provider supports t.38. i have a PSTN-fax available for testing. should i be able to send a fax to my asterisk and have it store it or send it by mail? can anyone point me to a extensions.conf |
| 07:02.24 | henk | example? |
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| 07:05.54 | l0st-soul | henk: i'm joining you on this question |
| 07:06.00 | l0st-soul | i'd like to know also |
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| 07:08.41 | schmidts | good morning |
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| 07:15.21 | Work2Play | wow good sized group here |
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| 08:10.30 | henk | can anyone tell me if anything happened with regards to this ticket: https://issues.asterisk.org/view.php?id=5177 ? it's about matching incoming calls correctly, when being registered to a provider multiple times. i have three sip accounts all with the same servername and -ip, asterisk always matched them to the last sip.conf-entry for that host and thus into the wrong context. |
| 08:15.25 | schmidts | henk not really, the easiest way would be to try out a trunk version if this problem still exists |
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| 08:28.32 | henk | schmidts: i'd rather not go through the hassle of building trunk without knowing if this issue is actually fixed there... any idea where to look that up? |
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| 08:40.18 | schmidts | henk chan_sip.c but i dont think you will find anything related very easy. you can take a look at the changes file maybe you will find something in there |
| 08:45.03 | henk | schmidts: mhm, ok, thank you |
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| 08:51.55 | wdoekes2 | henk: register with .../<myaccountname> and use match_auth_username=yes |
| 08:54.16 | wdoekes2 | (and then you may need to remove any insecure=invite lines you have.. matching by From would also work, but that depends on your provider not (ab)using it for callerid) |
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| 09:02.54 | atan | Inside the dialplan is there a way to not re-type the phone number each line? exten =>9055551212,1,Dial()... exten =>same_as_above or something? |
| 09:04.23 | henk | wdoekes2: ok, will have a look at that... i can't find any docs about that setting, do you know of one? |
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| 09:05.26 | tokozedg | hello, which is referred as B number, caller id or called? |
| 09:06.26 | WIMPy | atan: same => |
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| 09:17.24 | henk | wdoekes2: doesn't help, all calls still go to one peer... also match_auth_username is to find the _user_, not the peer, or is it? |
| 09:18.35 | henk | but since they are all type=friend which should equal a peer and a friend, it shouldn't matter, right? |
| 09:19.04 | wdoekes2 | henk: do a sip trace of an incoming call.. then we can see if it contains enough info to be matched to the right context |
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| 09:20.07 | wdoekes2 | I never fully understood the difference between users and peers.. I've only used friends ever |
| 09:20.10 | henk | wdoekes2: sip trace? sorry, never done that, what do i need? 'sip set debug on' is all i know for getting more info about it... |
| 09:20.28 | wdoekes2 | that one, yes |
| 09:20.38 | wdoekes2 | look for the packets beginning with INVITE |
| 09:22.52 | henk | http://pastie.org/1730234 |
| 09:23.52 | wdoekes2 | henk: the cli is in the From and no authentication is used |
| 09:24.10 | wdoekes2 | did you define a secret= ? |
| 09:24.16 | henk | wdoekes2: yes |
| 09:24.30 | wdoekes2 | and the insecure= option? |
| 09:25.00 | henk | invite |
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| 09:25.11 | wdoekes2 | 10:54 < wdoekes2> (and then you may need to remove any insecure=invite lines you have.. matching by From would also work, but that depends on your provider not (ab)using it for callerid) |
| 09:25.18 | Kanniball | hi! |
| 09:25.54 | wdoekes2 | we can see that the From is populated by the callerid, so you'll need auth-username matching for this to work |
| 09:25.55 | henk | invite= doesn't work, i get an error announcement about "user not being reachable" from my provider when calling asterisk |
| 09:26.48 | wdoekes2 | okay.. in that case you're out of luck, unless you can convince your provider to change some headers around |
| 09:27.33 | Kanniball | I have an setup, where an proprietary sip server forwards the calls through asterisk as go back to the server, although asterisk is making an invite changing the SDP. Is there any way to preserve the original SDP? |
| 09:28.05 | wdoekes2 | no Kanniball, asterisk is a b2bua.. it will proxy the rtp, so it needs to do sdp work |
| 09:28.09 | henk | mhm, ok, i'll just use one context and direct each incoming call to its own extension. imho it's pretty ugly and kind of sad that asterisk can't handle this situation :-/ |
| 09:28.41 | wdoekes2 | henk: what do you expect.. do you see anything in the sip packets you just captured that lets asterisk differentiate? |
| 09:28.54 | wdoekes2 | if you can't, don't call asterisk sad |
| 09:29.39 | Kanniball | wdoekes2: ok, but the problem is in the sdp, asterisk don't put the supported resolutions for video (which could make sense), but I need them |
| 09:29.40 | wdoekes2 | blame your provider.. the only way to work around this in another way, is to use different ports for the registrations.. but asterisk does not do that |
| 09:29.40 | henk | wdoekes2: To |
| 09:29.50 | henk | wdoekes2: INVITE |
| 09:29.58 | henk | wdoekes2: Contact |
| 09:30.11 | henk | wdoekes2: they all contain the username used for the peer... |
| 09:31.40 | wdoekes2 | hm |
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| 09:32.59 | wdoekes2 | Kanniball: if you just want the sip forwarded, you should probably use *SER (opensips, kamailio) |
| 09:35.06 | henk | wdoekes2: wrong? |
| 09:35.47 | wdoekes2 | henk: choosing account based on destination (R-URI and/or To) seems superfluous.. you would normally drop that in a single context and choose where to go from there (not ugly) |
| 09:36.23 | wdoekes2 | but I can see your point.. |
| 09:36.39 | E-bola | Do anybody know how to debug: chan_alsa.c:457 alsa_read: Read error: Resource temporarily unavailable |
| 09:36.48 | Kanniball | wdoekes2: but I already had one instance of asterisk (1.4 no I'm using 1.6), with this setup, but was a setup by an old collegue, and I don't have the configs. Is there any way do add the supported video resolutions to the sdp. |
| 09:36.52 | E-bola | I can play files with no problems via aplay |
| 09:37.30 | Kanniball | E-bola: just a wild guess: permissions on device or the device is already been using (if you don't have mixing support) |
| 09:37.58 | wdoekes2 | Kanniball: probably you can relay the video just fine.. but I don't know anything about that. look at configs/sip.conf.sample |
| 09:38.23 | E-bola | Kanniball: i've tried chmod'ing /dev/snd and /dev/audio1 to 777 |
| 09:38.39 | E-bola | and other utils like mocp and aplay can play audio fine... |
| 09:39.07 | Kanniball | E-bola: have you tried with the same user asterisk runs? |
| 09:39.10 | wdoekes2 | henk: see the [peer]-definition as a gateway entry point and the R-URI as an actual destination. then differentiating between the destinations from a single context may seem lees ugly to you |
| 09:39.47 | E-bola | Kanniball: to just su to asterisk (my asterisk runs as user asterisk ) ? |
| 09:40.16 | Kanniball | E-bola: yes, and try to play a file |
| 09:41.09 | E-bola | Kanniball: Yes that works fine, although im not sure its working, since asterisk has /bin/false as shell |
| 09:41.22 | wdoekes2 | (e-bola, kanniball: I don't know when playing a file would produce a *read* error..) |
| 09:41.28 | henk | wdoekes2: i see there are ways around it. still, i think matching incoming calls to peers _only_ based on the ip-address is really a bit weak... |
| 09:42.04 | Kanniball | wdoekes2: E-bola: you're right |
| 09:42.20 | E-bola | wdoekes2: Any idea what else might be wrong? |
| 09:42.28 | E-bola | i set input and output device to default |
| 09:42.43 | henk | not saying that's entirely asterisk's fault, since i have no clue about how the protocols work exactly, so it might be a deficit of sip... |
| 09:42.44 | E-bola | and as i said aplay works without changing any card or device options so i dont see why it wouldnt work |
| 09:43.15 | wdoekes2 | henk: it's not only.. it could differentiate on the port, on the From-user and on the authentication username |
| 09:43.28 | wdoekes2 | it's just that your peer does not differ on either of *those* |
| 09:43.50 | henk | wdoekes2: peer = sip provider in that case? or my peer declaration? |
| 09:43.59 | wdoekes2 | sip provider |
| 09:44.44 | E-bola | running aplay or mocp as user asterisk works fine |
| 09:44.49 | E-bola | so i dont think its a permissions issue |
| 09:45.04 | henk | so you're basically saying: if the sip provider doesn't do it, asterisk is not able to differentiate between two sip registrations to the same host? o_O |
| 09:45.04 | wdoekes2 | E-bola: you should check the source for clues |
| 09:45.16 | henk | is that different for tcp than for udp? |
| 09:45.27 | henk | because that sounds _really_ weird to me now... |
| 09:45.30 | Kanniball | E-bola: you have an SELinux enable system you could run into issues |
| 09:46.11 | E-bola | Kanniball: Hmm well im running debian squeeze, but i never looked deeply into SELinux, why would it be an issue? |
| 09:46.30 | schmidts | Does anyone of you remeber a problem with Subscribes and Notifys which are sent to a diferent ip + port then the original subscribe was received? |
| 09:46.50 | wdoekes2 | henk: I'm saying that normally you would differentiate the contact (in this case the sip provider) by From header.. but the provider puts the caller-id there |
| 09:47.28 | schmidts | my problem is that i see an incoming subscribe form my proxy and asterisk sends the 200 ok and also the initial notify back to the proxy, but the next notify is sent to the contact adress of the subscribe paket instead to the proxy |
| 09:48.18 | wdoekes2 | henk: it may very well be different for tcp because that uses a single open tcp connection (I think?).. that could allow asterisk to match the traffic, but it could be that it doesn't ;) |
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| 09:48.45 | E-bola | wdoekes2: mm found the place in the alsa channel code where the error comes from, but i dont have anywhere near enough insight to answer what the problem is :( |
| 09:49.20 | henk | wdoekes2: yes, but the consequence of that is that asterisk can't differ between two registrations to the same host and just chooses the first with a matching ip... sounds like really bad software and/or protocol design, doesn't it? |
| 09:50.07 | wdoekes2 | to asterisk, the register=> line is completely detached from the peer definition |
| 09:50.14 | schmidts | henk its about protocol design ;) |
| 09:50.48 | jamicque | hi @ll, I have a question. I have two simillar servers running only meetme, I've noticed that one has much more load than other. I've checked the number and te size of conferences, and they are more less the same. The codec used is only g711 (so no transcoding). Configurations of Asteriska are the same. Can somebody give me any hint? |
| 09:51.23 | wdoekes2 | the register simply tells the peer (sip provider) where you can be found.. the peer then calls you on your location (your_ip:5060).. and then it has to match for whom the call was |
| 09:52.25 | wdoekes2 | for sip trunks, the R-URI usually contains a destination telephone number (not an account code), so you wouldn't want to match the peer based on that |
| 09:52.28 | henk | oh, i have a /29, maybe i could tell asterisk to register to the first account using ip1, second using ip2 and so on? |
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| 09:52.55 | henk | let's see... |
| 09:53.06 | wdoekes2 | E-bola: can't help you with alsa, sorry |
| 09:53.27 | wdoekes2 | henk: I doubt it.. asterisk does not play well with multiple IPs |
| 09:53.38 | henk | argh |
| 09:54.02 | wdoekes2 | why do you dislike separating the destination based on extension in the dialplan? |
| 09:56.05 | wdoekes2 | one [sip-provider] peer context in your sip.conf, three register=> lines, and a single context in your dialplan |
| 09:57.03 | henk | wdoekes2: no specific reason so far, just my idea of how to configure asterisk: have every sip-account have its own context. i find that cleaner and it makes sense when reading and is easy to understand and clearly structured... |
| 09:57.07 | Kanniball | E-bola: I've just rememberd to add rules for asterisk under fedora, because I've got permission denied for lot of operations. |
| 09:59.18 | henk | wdoekes2: if all sip-accounts were used for similar purposes, i'd probably not care that much, but the rates for all of those are pretty differing and the supported features (t.38 for example) as well, so i use some for calling out, some for being called, some for fax, some for international calls, ... i need a clean structured config so i still have a good overview over what's going on. |
| 10:00.39 | E-bola | Kanniball: for devices or? |
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| 10:04.26 | wdoekes2 | henk: you could add a sip proxy (*SER) in the middle where you rewrite from-headers.. but that feels like too much work for too little gain |
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| 10:05.02 | henk | ack... anyway, lunch |
| 10:05.15 | henk | wdoekes2: thank you very much! you've been very helpful :) |
| 10:18.45 | *** part/#asterisk tokozedg (~torne@mail.orient-logic.com) |
| 10:26.28 | E-bola | lol what a pain |
| 10:26.57 | E-bola | after 3 hours i find out the problem isnt the console driver, but that asterisk for some reason isnt playing any ringing noise over the console, once i picked up the calls i could hear my voice fine over the speakers.... |
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| 10:45.21 | Kanniball | after reading at the mailling lists, I've found that I have the video codecs (format_h263) but I have no video codecs in core show translation. Any hint? |
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| 11:38.33 | StaRetji | orn: hi orn, remember my problem with mobile and master asterisk, I posted few days ago. I'm having problem where master doesn't always see mobile asterisk as registered, while mobile asterisk can make phone calls. |
| 11:38.58 | orn | yes |
| 11:39.01 | StaRetji | any ideas? |
| 11:39.39 | StaRetji | if I try to call mobile asterisk peer, it will succeed after I try several times |
| 11:39.56 | orn | well, for one, the box doesn't really need to be registered to make calls -- it just needs to send the correct authentication |
| 11:40.08 | StaRetji | ah, you're right |
| 11:40.11 | orn | but, can you make a sip capture of the failing and the succeeding calls and upload somewhere |
| 11:40.41 | StaRetji | is there a way to force mobile asterisk to check/ping/register or something every few seconds |
| 11:41.11 | StaRetji | because if I make a call from mobile asterisk, it is immediately reachable to other peers |
| 11:41.24 | StaRetji | after some time, it becomes unreachable |
| 11:41.37 | orn | could be a nat issue |
| 11:41.41 | StaRetji | yes |
| 11:41.45 | orn | but you could try qualify=yes in sip.conf |
| 11:42.00 | StaRetji | of mobile asterisk? |
| 11:42.01 | orn | that'll send an OPTIONS message every now and then to check the response time |
| 11:42.08 | orn | yes, or both |
| 11:42.09 | StaRetji | or I should set that in master sip.conf |
| 11:42.23 | StaRetji | on both, got it |
| 11:42.28 | orn | try that, see what happens |
| 11:42.42 | StaRetji | thx for the tip orn, I really appreciate your help |
| 11:42.55 | orn | potential side-effect: if it becomes unreachable or too lagged, the calls won't be sent to the peer |
| 11:42.57 | orn | no problem |
| 11:44.18 | StaRetji | oh, it's already qualify=yes at mobile asterisk |
| 11:44.21 | StaRetji | let me check master now |
| 11:44.24 | orn | ok |
| 11:46.16 | StaRetji | on master is qualify=1000 ( p3nguin suggestion ) was qualify = 8000 before that |
| 11:46.27 | StaRetji | I'm changing to qualify = yes now and will see |
| 11:46.33 | orn | you can change it to a higher value as well |
| 11:46.58 | orn | 1000 means that if the response time exceeds 1 second, calls will not be routed to it |
| 11:50.37 | StaRetji | once i reload configuration mobile peer is reachable, will see how it goes :) |
| 11:50.41 | StaRetji | thx dude ;) |
| 11:55.13 | StaRetji | nope, expired |
| 11:55.22 | StaRetji | second try, rings |
| 11:56.14 | StaRetji | I'm getting plenty of these in master cli port 5060 expires 120 |
| 11:56.27 | StaRetji | and it writes mobile asterisk IP |
| 11:56.52 | StaRetji | and port 1025 expires 120 |
| 12:00.41 | StaRetji | hah! |
| 12:01.18 | StaRetji | I've noticed once master asterisk shows port 5060 expires 120 mobile peers becomes reachable and can make call |
| 12:02.13 | orn | can you paste some of this output to pastebin ? |
| 12:02.51 | StaRetji | yes, one moment please |
| 12:06.55 | StaRetji | sorry, was on phone |
| 12:07.01 | StaRetji | will pastebin now |
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| 12:12.07 | StaRetji | orn: here it is mate |
| 12:12.41 | orn | where? |
| 12:12.42 | orn | :) |
| 12:12.46 | StaRetji | If you take a closer look on line 41 |
| 12:12.48 | StaRetji | lol |
| 12:12.55 | StaRetji | sorry http://pastebin.com/T0KFrAG6 |
| 12:13.20 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 12:13.23 | StaRetji | you will see that once I see port 5060 expires 120 |
| 12:13.29 | StaRetji | I'm able to make phone call |
| 12:13.36 | orn | ok |
| 12:13.42 | orn | this is a call to the mobile asterisk? |
| 12:13.47 | StaRetji | exactly |
| 12:14.20 | StaRetji | port 1025 expires 120, when this show up, I'm still not able to call, but once I see 5060, I try, call goes trough |
| 12:14.40 | StaRetji | so, calling mobile peer 7011 is lottery lol |
| 12:14.51 | orn | ok, that seems to indicate that the registration has expired |
| 12:15.09 | orn | <PROTECTED> |
| 12:15.16 | orn | that means that he just registered again |
| 12:15.21 | StaRetji | got it |
| 12:15.21 | orn | after which you are able to make calls |
| 12:15.26 | StaRetji | exactly |
| 12:15.33 | orn | have you tried setting the registration timeout higher? |
| 12:15.41 | StaRetji | and 120 seconds later maybe I can't, didn't count |
| 12:15.47 | StaRetji | how? |
| 12:16.07 | StaRetji | I mean, how can I do that? |
| 12:16.36 | orn | hmmm |
| 12:17.01 | orn | defaultexpiry=3600 |
| 12:17.03 | orn | for example |
| 12:17.06 | orn | that should set it to 1 hour |
| 12:18.26 | StaRetji | is this should be set on sip.conf of mobile asterisk? |
| 12:18.34 | orn | no, on the fixed asterisk |
| 12:18.39 | StaRetji | oh, ok |
| 12:18.43 | StaRetji | just found this |
| 12:18.44 | StaRetji | http://forums.whirlpool.net.au/archive/856483 |
| 12:18.48 | StaRetji | might be useful |
| 12:18.59 | orn | this needs to be in the [general] section as well |
| 12:19.37 | *** join/#asterisk Sipster (~Sipster@modemcable143.199-202-24.mc.videotron.ca) |
| 12:20.08 | orn | that is, it needs to be in the [general] section, not the section for this trunk |
| 12:20.36 | StaRetji | what will happen when I put defaultexpiry=3600 |
| 12:20.54 | StaRetji | it will consider mobile being online/registered at that IP for 1 hour? |
| 12:22.44 | *** join/#asterisk skrusty (~ben@83.166.169.221) |
| 12:22.47 | skrusty | afternoon all |
| 12:23.25 | *** join/#asterisk Godfather_ (~estanteri@164.Red-88-9-182.dynamicIP.rima-tde.net) |
| 12:25.16 | orn | StaRetji: yes |
| 12:25.38 | orn | StaRetji: unless the mobile re-registers |
| 12:26.00 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 12:31.17 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:37.15 | StaRetji | orn, that's great, will try it and let you know how it goes ;) |
| 12:45.14 | *** join/#asterisk Tuju (~tuju@84-50-207-175-dsl.est.estpak.ee) |
| 12:48.11 | *** join/#asterisk coppice (~chatzilla@9.160.232.220.dyn.pacific.net.hk) |
| 12:55.28 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 12:57.53 | StaRetji | orn, same thing :/ |
| 12:58.34 | orn | after the same amount of time? |
| 12:58.52 | StaRetji | well, not sure, I called aster I applied changes |
| 12:58.59 | StaRetji | went for a cigarette break |
| 12:59.02 | orn | ok |
| 12:59.06 | StaRetji | tried again, nothing |
| 12:59.13 | orn | are you having the fixed asterisk register with the mobile asterisk as well? |
| 12:59.25 | StaRetji | no, not possible |
| 12:59.26 | orn | if so, that might be the problem, and might be unnecessary since its ip address never changes |
| 12:59.31 | StaRetji | don't know ip of mobile |
| 12:59.41 | orn | ok |
| 12:59.52 | orn | on the mobile asterisk, try setting insecure=very on the fixed trunk |
| 12:59.53 | StaRetji | sometimes is this one, sometimes other one |
| 13:00.30 | saxa | hi, anybody knows why I always get only "asterisk" on my phones instead of the caller number ? |
| 13:00.44 | StaRetji | orn, not sure if it can work, mobile has asterisk 1.8 |
| 13:00.46 | *** join/#asterisk Freeaqingme (~dolf@dsl-083-247-011-232.solcon.nl) |
| 13:00.51 | StaRetji | master is 1.4 |
| 13:00.57 | saxa | I tried all of the possible settings in dahdi_chan.conf |
| 13:00.57 | StaRetji | will try though |
| 13:01.53 | orn | StaRetji: Has insecure been removed in 1.8 ? |
| 13:03.03 | StaRetji | well, i trued insecure=very before and it didn't work |
| 13:03.07 | StaRetji | let me check now again |
| 13:03.18 | orn | mind you, do it on the mobile asterisk |
| 13:03.23 | StaRetji | I mean, it threw errors I can't remember |
| 13:03.23 | StaRetji | ok |
| 13:04.07 | StaRetji | Failed to authenticate device "7777" <sip: |
| 13:04.10 | StaRetji | so, it wont do |
| 13:04.16 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:04.17 | orn | did you do sip reload? |
| 13:04.23 | StaRetji | but it reaches mobile asterisk |
| 13:04.24 | StaRetji | yes |
| 13:04.31 | StaRetji | which is good |
| 13:04.46 | orn | can you put your sip.conf on the mobile asterisk on pastebin? |
| 13:04.57 | StaRetji | yes, give me a minute |
| 13:05.28 | orn | StaRetji: I believe it always reached the mobile asterisk, as the other one was receiving forbidden |
| 13:07.08 | StaRetji | orn: http://pastebin.com/rbKZSSB2 |
| 13:08.34 | orn | ok, and the corresponding config from the master server? |
| 13:08.45 | StaRetji | ok, second |
| 13:10.45 | orn | on mobile, try setting insecure to insecure=port,invite |
| 13:10.48 | StaRetji | http://pastebin.com/a1b3FQhz |
| 13:11.05 | StaRetji | insecure=port,invite |
| 13:11.19 | StaRetji | it was like that before you said to put very |
| 13:11.29 | StaRetji | changing it back... |
| 13:12.12 | orn | oh ok |
| 13:12.54 | orn | can you also paste the output from the mobile asterisk when you make a call from the other one and it fails? |
| 13:13.44 | StaRetji | there is nothing |
| 13:13.52 | orn | at verbosity level 9? |
| 13:13.56 | StaRetji | cli doesn't throw anything |
| 13:13.58 | StaRetji | level 15 |
| 13:14.07 | StaRetji | so, I assume it never reches it |
| 13:14.16 | orn | the master receives "forbidden" |
| 13:14.21 | orn | so it's talking to some sip server for sure |
| 13:14.39 | orn | run tcpdump on the mobile asterisk when you make a call and see if you see it there |
| 13:15.06 | StaRetji | let me try again, to be sure |
| 13:15.26 | StaRetji | give me few minutes to become unreacheable |
| 13:19.00 | *** join/#asterisk m4xmr (~m4xmr@93-36-129-84.ip60.fastwebnet.it) |
| 13:19.10 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:20.24 | jaytee | how can I block a specific ip address from trying to register to my asterisk pbx in iptables. I don't have Fail2Ban installed (yet) and need to block it now until I can get Fail2Ban setup and tested. |
| 13:21.15 | jaytee | i tried iptables -A INPUT -s xxx.xxx.xxx.xxx -j DROP but I'm still getting brute force registration attempts from that IP |
| 13:23.40 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:23.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:24.39 | kleszcz | try echo "ALL:IP" >> /etc/hosts.deny |
| 13:26.46 | kleszcz | if u wan i can give you configuration for fail2ban |
| 13:27.06 | StaRetji | orn, to confirm, no reaction in CLI when no reachable |
| 13:27.17 | orn | ok, you need to do a tcpdump then |
| 13:31.24 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 13:32.43 | Aut0ExeC | hi guys... small question for you... if I have disconnect supervision(aka cut off on disconnect) on my pstn line but my line has a lot of static... could that cause asterisk to not drop the call after the Hangup? |
| 13:32.43 | StaRetji | orn: nothing, just SSH port and my laptop IP |
| 13:32.55 | StaRetji | so, it seems it never reaches mobile asterisk |
| 13:33.05 | Aut0ExeC | I'm using cisco spa3102 |
| 13:33.11 | orn | StaRetji: judging from the tcpdump? |
| 13:33.12 | StaRetji | who is giving forbidden is a mistery to me |
| 13:33.13 | vfabi | hi all , where to find realtime sipusers table fields description ? |
| 13:33.18 | StaRetji | StaRetji: yes |
| 13:33.28 | orn | StaRetji: Then do a tcpdump on the master and see where it's sending it to |
| 13:33.33 | StaRetji | right |
| 13:35.15 | *** join/#asterisk m4xmr (~m4xmr@93-36-129-84.ip60.fastwebnet.it) |
| 13:36.41 | kleszcz | jaytee: http://asterisk.pastebin.pl/39009 http://asterisk.pastebin.pl/39010 |
| 13:37.08 | seraphie | vfabi: http://ofps.oreilly.com/titles/9780596517342/ch16.html#I_section12_tt1465 |
| 13:38.08 | seraphie | vfabi: search on page for "sipusers realtime table" |
| 13:38.21 | seraphie | Table 16.3 |
| 13:38.59 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:39.18 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 13:39.30 | vfabi | <PROTECTED> |
| 13:39.40 | no1peanut | Hi, I want to playback audio into a call, as caller - for both parties to hear. Can I do this using dynamic_features ? |
| 13:41.05 | Aut0ExeC | do most of you guys integrate pstn lines into ur asterisk setup? |
| 13:41.57 | StaRetji | orn, by looking at master server tcpdump it looks like it tries mobile asterisk |
| 13:43.38 | StaRetji | but on mobile ast I don't see it |
| 13:43.51 | StaRetji | cli nor tcpdump |
| 13:49.05 | orn | then your service provider must be doing something |
| 13:49.07 | orn | or your router |
| 13:49.15 | orn | how is the mobile asterisk connected? |
| 13:49.26 | StaRetji | to adsl router |
| 13:49.34 | StaRetji | but I have sip phone connected directly to master |
| 13:49.43 | StaRetji | that one works without a problem |
| 13:49.58 | orn | my guess is that the SIP phone is receiving the INVITE in the cases where it fails |
| 13:50.11 | orn | router messing up the NAT |
| 13:50.32 | StaRetji | sounds reasonable |
| 13:50.46 | StaRetji | I will try checking router, local ip of mobile asterisk |
| 13:50.48 | orn | try registering either one on non-standard ports |
| 13:51.02 | StaRetji | for me, information that those conf files are okay is more than enough |
| 13:51.12 | StaRetji | got it |
| 13:53.05 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 13:54.24 | Aut0ExeC | you guys using pstn lines? |
| 13:54.32 | Aut0ExeC | ? |
| 13:56.14 | atan | So I get "check_auth: username mismatch, have <19055551212>, digest has <s>" when I try to name a SIP account the same as the phone number that's forwarding to it. Is it simply because I forgot to set something somewhere? |
| 13:59.40 | wdoekes2 | atan: register with register => .../<username> |
| 14:01.56 | StaRetji | orn, chaning port on my router did it! It seems that SIP phone and mobile asterisk connected to same public ip are not working properly if they use same port! |
| 14:02.39 | StaRetji | orn: thank you so much for helping out. I've wouldn't fix it if you didn't drive me thoroughly! |
| 14:02.45 | StaRetji | Thx dude :) |
| 14:05.37 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 14:07.16 | *** join/#asterisk massoud (~massoud@unaffiliated/massoud) |
| 14:11.12 | orn | StaRetji: no problem. glad to help. :) |
| 14:11.30 | massoud | ls |
| 14:11.37 | orn | . |
| 14:11.38 | orn | .. |
| 14:11.45 | massoud | :) |
| 14:12.05 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 14:12.52 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 14:17.02 | *** join/#asterisk Freeaqingme (~dolf@dsl-083-247-011-232.solcon.nl) |
| 14:22.30 | Kanniball | anybody knows how I do setup asterisk to announce supported video resolutions in SDP? |
| 14:25.32 | *** join/#asterisk zkn (~zkn@213.115.26.228) |
| 14:25.42 | jaytee | Kanniball, SDP? |
| 14:27.07 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:27.26 | *** join/#asterisk bobg (~bobg@ool-4576d9c2.dyn.optonline.net) |
| 14:29.29 | *** join/#asterisk GTXComm (~John@72.128.62.30) |
| 14:29.47 | bobg | i have a production asterisk installation and the owner of the company just emailed that he's on a call that has been dropped several times (he has to reconnect the call) |
| 14:29.57 | bobg | how can I debug this? |
| 14:30.18 | bobg | I don't see anything unussual in the logs |
| 14:32.22 | bobg | the call path is aastra phone --(SIP/ulaw)--> asterisk box 1 --(IAX/ulaw)--> asterisk box 2 --(SIP/ulaw)--> provider SIP |
| 14:32.30 | bobg | box 1 and box 2 are ours |
| 14:33.27 | Kanniball | jaytee: yes, in the invite that Asterisk sends, there's informatition about supported codecs, but in the video there's no info about the resolutions |
| 14:33.28 | bobg | any ideas of what I can do to monitor this call in progress so that I can understand what drops it if it happens again? |
| 14:33.51 | orn | bobg, turn on SIP debugging or use tcpdump to a file |
| 14:33.59 | *** join/#asterisk scud (c09758bc@gateway/web/freenode/ip.192.151.88.188) |
| 14:34.02 | orn | bobg, that way you can see who is sending the BYE message |
| 14:34.11 | scud | angler: whats up, support me bitch. |
| 14:34.30 | bobg | orn, thanks |
| 14:36.04 | bobg | orn, i did "sip set debug on" on the console -- that will cause th sip messages to be logged to the 'full' log, won't it? |
| 14:36.28 | orn | yes, should do, if the full log is configured |
| 14:36.32 | orn | it'll be a lot of messages though |
| 14:37.00 | orn | would probably be easier to use tcpdump to a file and open it with wireshark so you can easily find the call |
| 14:37.05 | bobg | yes, I will be sure to tunr it off after this call ends |
| 14:43.00 | bobg | now I am trying to identify the channels for this call... on the box1, the channel is easy to identify by his extension, but how can i find out which of the SIP channels on box 2 (going to the provider) this call is using? |
| 14:43.29 | orn | while it's ongoing? |
| 14:43.35 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:46.03 | *** join/#asterisk antoasla_ (~antoasla@athedsl-4551202.home.otenet.gr) |
| 14:46.11 | bobg | yes |
| 14:46.30 | bobg | the call is going on now |
| 14:47.01 | *** join/#asterisk nickfennell (~nick@cov1.appliansys.com) |
| 14:47.19 | bobg | oh. after the call end, the information in the cdr will help me piece it together, right? |
| 14:47.26 | nickfennell | hey guys |
| 14:47.38 | nickfennell | How do I instruct asterisk to only use g729 |
| 14:47.41 | nickfennell | disallow=all |
| 14:47.44 | nickfennell | allow=g729 ? |
| 14:48.12 | orn | correct |
| 14:48.32 | bobg | nickfennell, but you also have to make sure you put that in the right place |
| 14:49.07 | nickfennell | I'm trying it per trunk |
| 14:49.16 | nickfennell | My phone reports invalid when dialling a call |
| 14:49.25 | bobg | i.e. it can be at the global SIP level to set the default, but then it can be overridden for each trunk or user |
| 14:49.28 | nickfennell | how can I check asterisk has a valid g729 licence? |
| 14:49.52 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:50.07 | *** join/#asterisk dschuett (~dschuett@wsip-68-15-229-108.om.om.cox.net) |
| 14:50.18 | dschuett | does this look correct?: http://pastebin.com/nWuMZuwc |
| 14:50.43 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 14:51.18 | no1peanut | I want to playback audio into a call, as caller - for both parties to hear. Can I do this using dynamic_features ? |
| 14:53.08 | antoasla_ | hello all i have a small problem trying to configure a simple (3 softphones) network. I think i configured right the sip.conf and extensions.conf files but i can connect a softphone from my laptop witch is in the same network with the asterisk server. Any ideas? should i post the sip.conf and extensions.conf? |
| 14:53.23 | russellb | no1peanut: yes, but it's not obvious how. leifmadsen, don't we have a recipe for that? |
| 14:53.26 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:53.41 | dschuett | antoasla: yes, post your configs |
| 14:53.58 | antoasla_ | [phone1] ;1st phone client |
| 14:53.59 | antoasla_ | nat=yes |
| 14:54.01 | antoasla_ | type=friend |
| 14:54.02 | antoasla_ | host=dynamic |
| 14:54.04 | antoasla_ | username=phone1 |
| 14:54.05 | antoasla_ | secret=*** |
| 14:54.07 | antoasla_ | dtmfmode=inband ;Choices are inband, rfc2833, or info |
| 14:54.08 | antoasla_ | context=sip |
| 14:54.10 | antoasla_ | callgroup=1 |
| 14:54.10 | *** mode/#asterisk [+q antoasla_!*@*] by russellb |
| 14:54.13 | russellb | ~pb |
| 14:54.14 | 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.us , 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. |
| 14:54.16 | leifmadsen | russellb: the no1peanut one? ya I think so -- check the Audio Manipulation chapter |
| 14:54.21 | russellb | leifmadsen: yes |
| 14:54.31 | leifmadsen | ya there is a recipe for that exact thing I'm pretty sure |
| 14:54.31 | russellb | no1peanut: check out the asterisk cookbook of ofps.oreilly.com |
| 14:54.37 | russellb | yeah i thought so |
| 14:54.47 | russellb | we need ~cookbook |
| 14:54.54 | leifmadsen | ~cookbook |
| 14:54.54 | infobot | methinks cookbook is "Over 1,500 time-saving recipes and hints for busy modern computer users" at http://dsl.org/cookbook/ |
| 14:54.59 | leifmadsen | ~asteriskcookbook |
| 14:55.06 | leifmadsen | thought I added something about that a while ago |
| 14:55.17 | leifmadsen | perhaps not |
| 14:55.22 | leifmadsen | ~astcookbook |
| 14:55.22 | dschuett | anyone good with agi scripts? |
| 14:55.25 | no1peanut | russellb, I've been there and the recipe I found was only to play audio for the caller not both |
| 14:55.46 | leifmadsen | no1peanut: adjust the features.conf to play to both |
| 14:55.47 | russellb | it's using ChanSpy right? |
| 14:55.47 | no1peanut | russellb, I might have misunderstood though ;) |
| 14:55.58 | russellb | change ChanSpy() to use barge mode |
| 14:56.03 | russellb | and it will play to both |
| 14:56.07 | leifmadsen | \o/ |
| 14:56.09 | leifmadsen | solutions! |
| 14:57.25 | russellb | leifmadsen: if we solve enough problems maybe someone will buy the book |
| 14:57.29 | no1peanut | russellb, ahh .. *palm on forehead* |
| 14:57.52 | leifmadsen | russellb: that'd be the ideal :) |
| 14:58.35 | russellb | leifmadsen: 6 days to print on ATDG! |
| 14:58.56 | leifmadsen | russellb: that's amazing! I better get my thanks section updated :) |
| 14:59.24 | p3nguin | staretji, orn: As I stated before, in the case of setting qualify=yes, yes = 2000. 1000 is half the time of "yes". yes equals 2 seconds, 1000 equals 1 second. |
| 14:59.54 | *** part/#asterisk antoasla_ (~antoasla@athedsl-4551202.home.otenet.gr) |
| 15:00.00 | *** join/#asterisk antoasla_ (~antoasla@athedsl-4551202.home.otenet.gr) |
| 15:00.59 | no1peanut | russellb, leifmadsen - I will buy both ! And the one about writing modules ;) |
| 15:01.18 | leifmadsen | assigned 2 shiny nickles to a new toy |
| 15:01.24 | leifmadsen | assigns* |
| 15:01.25 | leifmadsen | heh |
| 15:02.58 | *** part/#asterisk dschuett (~dschuett@wsip-68-15-229-108.om.om.cox.net) |
| 15:09.41 | *** join/#asterisk jkroon (~jkroon@dsl-241-240-52.telkomadsl.co.za) |
| 15:10.47 | jkroon | hi all, other than target practice at the trashcan - does anybody have some experience with the grandstream FXS gateways? It has an "FXO Failover gateway" option which I'm trying to set up but it does not seem to want to work at all ... any ideas on where to start digging? |
| 15:11.17 | jkroon | the syslog messages aren't being particularly useful either and google has helped only in finding manuals which doesn't exactly help either. |
| 15:11.29 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.saucontech.com) |
| 15:11.46 | leifmadsen | russellb: can you unblock antoasla_ (unless there is a reason not to that I missed) |
| 15:12.09 | tzanger | he slept with russelb's wife |
| 15:12.14 | tzanger | oh wait no that was me |
| 15:12.42 | fullstop | Hi all. I'm having some trouble with STRPTIME.. http://pastebin.com/Cmqd5hTe |
| 15:13.15 | *** part/#asterisk scud (c09758bc@gateway/web/freenode/ip.192.151.88.188) |
| 15:13.15 | fullstop | That dialplan snippet returns with "[Mar 29 11:09:58] WARNING[15892]: func_strings.c:899 acf_strptime: STRPTIME() found no time specified within the string" |
| 15:13.31 | leifmadsen | tzanger: zing! |
| 15:13.42 | fullstop | Shouldn't strftime produce something which can be read by strptime? |
| 15:13.51 | fullstop | given the same format string, that is.. |
| 15:13.57 | leifmadsen | fullstop: you have % instead of a $ |
| 15:14.04 | leifmadsen | %{TIME_VAR} is wrong |
| 15:14.08 | leifmadsen | ${TIME_VAR} |
| 15:14.09 | fullstop | doh |
| 15:14.10 | fullstop | :-D |
| 15:14.12 | leifmadsen | :) |
| 15:14.14 | leifmadsen | \o/ |
| 15:14.18 | fullstop | Thanks for pointing out the obvious. :D |
| 15:14.30 | leifmadsen | it bites us all at least once :) |
| 15:15.19 | fullstop | leifmadsen: Unfortunately, that didn't correct my problem. :-/ |
| 15:15.26 | *** join/#asterisk sjobeck (~sjobeck@206.83.218.66) |
| 15:15.44 | fullstop | I get the same result if I pass in a literal "6:55 am" string. |
| 15:16.39 | *** part/#asterisk sjobeck (~sjobeck@206.83.218.66) |
| 15:17.04 | fullstop | The documentation for strptime is pretty sparse |
| 15:18.15 | _Raptor_ | manual says i can get a list of channels with $CHANNELS(SIP/provider). this means, if 3 people use the provider at the same time, it results 3 channels. how can i iterate through this list and do stuff |
| 15:18.40 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 15:18.55 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.140.123) |
| 15:18.56 | _Raptor_ | or how can i get the channel which was created by Dial(SIP/provider/12345)? |
| 15:19.46 | leifmadsen | _Raptor_: it sounds more like you want to use GROUP() and GROUP_COUNT() to count the number of channels |
| 15:19.55 | leifmadsen | runs off to get cofee |
| 15:19.57 | leifmadsen | coffee* |
| 15:20.51 | _Raptor_ | leifmadsen: i don't need to count the chans, i just need to find the chan which called a specific number. but i will have a look at GROUP |
| 15:20.57 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 15:21.41 | ujjain | I am wondering if my Asterisk succesfully connects to SIP-trunk, it keeps on retransmitting (http://www.codepad.eu/view/45894658), sip show peers shows online though |
| 15:23.49 | _Raptor_ | leifmadsen: what i am trying to do is sth like that: for chan in $CHANNELS(SIP/provider); do if ($chan(dialed_number) == 1234) then ... |
| 15:26.55 | nickfennell | is it possible to make asterisk re-register sip trunks? |
| 15:31.43 | orn | nickfennell: sip reload |
| 15:31.56 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 15:33.10 | nickfennell | that's what I thought |
| 15:33.11 | nickfennell | thanks |
| 15:37.16 | fullstop | I don't think that STRPTIME works with AM and PM |
| 15:40.34 | *** join/#asterisk marienz (~marienz@freenode/staff/marienz) |
| 15:47.46 | ujjain | Retransmitting #3 (NAT) to 83.143.188.165:5060 many times: < does this mean registration is failing? |
| 15:48.13 | ujjain | Via: SIP/2.0/UDP 192.168.1.116:5060;branch=z9hG4bK783f88ad;rport / Call-ID: 4f6d1b7d2a690e35235e6c8375430646@192.168.1.116 |
| 15:48.14 | ujjain | internal IP... while NAT=yes... |
| 15:52.47 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:55.16 | *** join/#asterisk Tuju (~tuju@84-50-207-175-dsl.est.estpak.ee) |
| 15:59.02 | *** join/#asterisk Schreiber1337 (cee4b465@gateway/web/freenode/ip.206.228.180.101) |
| 16:00.15 | Schreiber1337 | Anyone here ever use a SPA505S sidecar? |
| 16:01.51 | Schreiber1337 | The connection string uses $exten@$IP.... but how do you specify the secret in this string? |
| 16:02.39 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 16:02.49 | Qwell | if it's not stupid, you probably do $exten:password@IP |
| 16:03.40 | nickfennell | If i want to change the SIP port |
| 16:03.45 | nickfennell | just port=5065 ? |
| 16:03.57 | Qwell | nickfennell: the port Asterisk binds to, you mean? |
| 16:04.01 | nickfennell | yes |
| 16:04.04 | Qwell | bindport= |
| 16:04.32 | nickfennell | in sip.conf ? |
| 16:04.38 | Qwell | yes |
| 16:04.51 | *** join/#asterisk tstorm (~tstorm@173-164-230-21-SFBA.hfc.comcastbusiness.net) |
| 16:06.10 | nickfennell | Will this effect where phones register to ? |
| 16:06.23 | Qwell | You have to modify the phones to use that port. |
| 16:06.42 | nickfennell | So everything must use 5065 ? |
| 16:06.54 | *** join/#asterisk mbowie (~mbowie@173.228.121.4) |
| 16:06.56 | Qwell | That would be logical, yes. |
| 16:07.21 | Qwell | They don't have to listen on that port - but they have to know where to connect. |
| 16:07.51 | nickfennell | Can i use 5060 internally |
| 16:07.59 | nickfennell | but then use 5065 for outside access ? |
| 16:08.08 | Qwell | Asterisk can only listen on one port. |
| 16:08.18 | Qwell | You could possibly use some iptables magic, but it's not trivial. |
| 16:08.50 | nickfennell | hmm ok |
| 16:08.51 | Qwell | Or you could put a proxy in front of Asterisk to the outside. |
| 16:10.57 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 16:11.35 | mbowie | Good morning folks... possibly a noob question, but I have a new Asterisk 1.8.2.3 setup with a SIP trunk through Broadvox and am having a struggle with sending a callerId number via the trunk. They accept the "From:" header for this purpose, but my config is sending "Unknown". |
| 16:12.35 | mbowie | I've tried setting sendrpid for the trunk, callerid, fromuser and cid_number for the "extension". |
| 16:14.28 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 16:16.59 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 16:23.08 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 16:23.34 | *** part/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 16:25.53 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 16:27.04 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:1015:f915:f4f7:fcaa) |
| 16:29.45 | ujjain | Can somebody please help me find out why my incoming calls fail? http://www.codepad.eu/view/71281138 |
| 16:29.45 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 16:31.10 | Qwell | you don't have an extension for 31107142866? |
| 16:31.22 | nickfennell | hmmm, common causes for a 603 ? |
| 16:31.46 | ujjain | Qwell: the extension is 101. |
| 16:32.29 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 16:33.00 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 16:33.00 | ujjain | Qwell: I have a trunk for 31107142866, budgetphone trunk. |
| 16:33.04 | *** join/#asterisk mountainm2k (~msturtz@gw.booyahnetworks.com) |
| 16:33.45 | p3nguin | Asterisk doesn't recognize "trunks." It recognizes peers/users/friends and extensions. |
| 16:33.46 | retentiveboy | What's the right way to configure/handle SIP reinvites when the PBX has both LAN and WAN interfaces? Having trouble when public SIP clients call internal stations. |
| 16:33.46 | mountainm2k | Are Digium FXS modules compatible between 400 and 410 boards? |
| 16:34.10 | Qwell | retentiveboy: use externip and such |
| 16:34.55 | retentiveboy | Qwell: thought that only applied when NAT'ing |
| 16:35.11 | ujjain | I have configured the trunks as peers, and the 101-extension as friend, I set up incoming calls to all go to extension 101. |
| 16:35.38 | ujjain | I am using NAT, unlike on my previously working Asterisk server which I copied settings from. I did add nat=yes, qualify=yes. |
| 16:35.57 | p3nguin | externip (externaddr) or externhost is used for any Asterisk which is connected to both a LAN and a WAN. |
| 16:36.20 | mountainm2k | I have a TDM400 board, and one FXS, and I need a second. Oddly, I have three FXO's that I don't need. Seems like they don't make or support the 400 board anymore, but wondering if I can buy the 410 FXS modules... |
| 16:36.24 | p3nguin | It doesn't matter if the host where Asterisk resides is the NAT server or not. |
| 16:36.50 | Qwell | mountainm2k: If you're unsure, call Digium sales. That's why they're there. |
| 16:36.59 | ujjain | I have configured externIP ¨WanIP¨. |
| 16:37.13 | ujjain | the ports are forwarded from externIP to internIP. |
| 16:37.36 | mountainm2k | Qwell: Thanks, I had hoped to avoid that :-) |
| 16:40.25 | *** join/#asterisk Schreiber1337 (cee4b465@gateway/web/freenode/ip.206.228.180.101) |
| 16:40.28 | *** join/#asterisk b14ck_ (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 16:41.09 | Schreiber1337 | @Qwell: I guess the Cisco SPA505S is stupid... I still get a failed connection |
| 16:41.59 | Schreiber1337 | Does anyone have any experience working with any Cisco or Linksys sidecars? |
| 16:44.23 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-ypxmztbhwkpmkqci) |
| 16:46.31 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 16:49.34 | ruben23 | hi guys how do i handle voicemail when teh password have numeric and letters, whihc when i dial the phone it wotn recognized at all |
| 16:51.38 | p3nguin | How did the password get letters in it? Access it in the same way they were put there to begin with. |
| 16:52.00 | *** part/#asterisk mountainm2k (~msturtz@gw.booyahnetworks.com) |
| 16:52.30 | p3nguin | If you put them there in the config, change it in the config to get the letters out. |
| 16:54.00 | retentiveboy | I've adjusted sip.conf to set localnet maching my LAN and externaddr matching my WAN address. still getting remotely bridged calls. Grrr. |
| 16:54.26 | p3nguin | What does "getting remotely bridged calls" mean? |
| 16:56.03 | retentiveboy | external caller connects via SIP URL to to my WAN interface, dialplan rings internal station, answer the call on the station. Console says "Remotely bridging SIP/... and SIP/..." and I get no audio. |
| 16:57.23 | *** join/#asterisk imox1234 (~imox1234@p4FC5C791.dip0.t-ipconnect.de) |
| 16:59.57 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 17:00.00 | *** part/#asterisk StaRetji (~BigAll@91.143.222.166) |
| 17:00.29 | p3nguin | Did you set canreinvite/directmedia to no for the offending peers? |
| 17:00.45 | p3nguin | Also set it as a general setting so anonymous peers are affected. |
| 17:00.51 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v011-018.mobile.uci.edu) |
| 17:01.00 | p3nguin | s/peers/callers/ |
| 17:01.27 | p3nguin | Keep Asterisk in the media path. |
| 17:01.31 | retentiveboy | Thought I'd tried that. Will check. |
| 17:02.15 | *** join/#asterisk raden_work (~jon@66-191-96-74.static.eucl.wi.charter.com) |
| 17:03.02 | mbowie | Good morning folks... possibly a noob question, but I have a new Asterisk 1.8.2.3 setup with a SIP PSTN gateway through Broadvox and am having a struggle with sending a callerId number via the trunk. They accept the "From:" header for this purpose, but my config is sending "Unknown". |
| 17:04.57 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 17:05.43 | retentiveboy | p3nguin: directmedia=no seems to work for this case. now to see what that change has broken :) thx |
| 17:05.45 | p3nguin | While on this topic, if I want to use canreinviate=nonat, do I need to first set canreinvite=no and then canreinvite=nonat too? Or just use the nonat setting? |
| 17:06.27 | p3nguin | Setting directmedia=no as a general option shouldn't break anything. |
| 17:07.03 | p3nguin | The worst it will do is keep asterisk in the media path for peers that you haven't explicitly defined a directmedia setting for. |
| 17:07.21 | ruben23 | hi guys how do i change the voicemail voice prompt..? |
| 17:07.23 | retentiveboy | yeah, wondering how many stations I need to manually add that to now |
| 17:07.57 | Work2Play_ | this place seems to get moving during the day! |
| 17:08.07 | p3nguin | If I can get an answer about the nonat value, that could be something you might want to consider. |
| 17:08.22 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 17:08.52 | *** join/#asterisk cjk_ (~cjk@85.93.217.128) |
| 17:09.01 | cjk_ | hi, does anyone know this msg? Could not locate a FAX technology module with capabilities (RECEIVE) |
| 17:16.38 | retentiveboy | p3nguin: looking in the code, I see a coupel things. directmedia and canreinvite appere to be synonymous. The value can be true or false or a combination of update and nonat. Doesn't looke like no,nonat would fly |
| 17:17.21 | retentiveboy | also looks like multiple sets of the field would be superfluous; last one wins. |
| 17:25.53 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 17:26.13 | *** join/#asterisk josta (~josta@unaffiliated/josta) |
| 17:26.24 | p3nguin | Perfect. |
| 17:26.40 | p3nguin | So now you can consider the nonat option as well. |
| 17:29.09 | *** join/#asterisk Charrit (~Zairus@226.109.165.83.dynamic.mundo-r.com) |
| 17:29.16 | retentiveboy | tried setting directrtp=nonat globlly along with externaddr and localnet. didn't work |
| 17:29.36 | p3nguin | hmm |
| 17:29.38 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:29.48 | retentiveboy | exactly, hmmm |
| 17:31.57 | p3nguin | It sounded like nonat would work for your situation. |
| 17:32.10 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 17:32.46 | retentiveboy | gonna take a step back and read some code. I'd not expect this topology to be uncommon. |
| 17:32.55 | *** join/#asterisk torgnyw (~torgnyw@85.19.215.43) |
| 17:35.03 | cjk_ | Hi, can anyone tell me how to allow FAX on SIP with G711. At the moment I get "Audio FAX not allowed on channel" |
| 17:37.27 | ectospasm | cjk_: not a good idea, transmitting fax with G.711 over SIP is unreliable. |
| 17:37.47 | ectospasm | use T.38 |
| 17:39.43 | *** join/#asterisk newsham (~chat@udp216902uds.hawaiiantel.net) |
| 17:39.50 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 17:40.04 | newsham | hi. I had asterisk configured to work with googvoice/gtalk and it was working fine, but now its not working. |
| 17:40.24 | newsham | is it working for others? did I mess something up or did something change? |
| 17:41.17 | newsham | I see the jabber presence, but i cant even start a gtalk chat with the account anymore |
| 17:41.52 | cjk_ | ectospasm, well this isnt really the question to my answer |
| 17:42.05 | cjk_ | ectospasm, there are scenarios where passthrough is by far a better idea |
| 17:49.17 | Charrit | Can I manage Asterisk CDR to store the reports in this file schema? |
| 17:49.26 | Charrit | Master_YYYYMMDD.cvs |
| 17:49.56 | Charrit | or should I parse manually the Master.csv to split it into several files? |
| 17:53.31 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:53.41 | fauxalliance | logrotate Charrit |
| 17:54.19 | Charrit | yes, but I was looking for something more acurate, with logrotate I can miss some calls in CDR |
| 17:54.33 | Charrit | until I make the logger reload |
| 17:55.05 | fauxalliance | potentially |
| 17:55.15 | Charrit | logrotate is a good approach for logs (I already use it) but not for CDR |
| 17:56.06 | fauxalliance | i prefer the relational database approach personally |
| 17:56.30 | *** join/#asterisk cerberus_za (~coert@196-215-119-193.dynamic.isadsl.co.za) |
| 17:56.44 | Charrit | yes, me too, and I'm going to do it but in assyncronous maner at the end of the day |
| 17:57.04 | fauxalliance | Charrit, ideal :) |
| 17:57.04 | Charrit | If I make the cdr_mysql directly I can overload the server |
| 17:57.20 | fauxalliance | well, sounds lke a job for another node. |
| 17:59.16 | Charrit | perhaps, I expected that asterisk can manage it with cdr_custom |
| 18:01.53 | newsham | ok, the goog problem was definitely my fault, at least for incoming voice calls. |
| 18:02.25 | newsham | for some reason the jabber presence isnt working as well as it as the other day and i cant call from gtalk direclty, but i dont care too much about that |
| 18:02.35 | newsham | i can at least make incoming calls again :) |
| 18:03.32 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 18:04.29 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 18:04.39 | *** join/#asterisk bklang (~bklang@2001:470:8:5de:21f:f3ff:fed8:6a5d) |
| 18:05.14 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 18:06.02 | *** join/#asterisk imox1234 (~imox1234@p4FC5C791.dip0.t-ipconnect.de) |
| 18:09.15 | *** part/#asterisk newsham (~chat@udp216902uds.hawaiiantel.net) |
| 18:22.55 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 18:33.20 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 18:34.30 | *** join/#asterisk oej (~olle@95.209.191.71.bredband.tre.se) |
| 18:35.51 | cjk_ | Hi, can anyone tell me how to allow FAX on SIP with G711. At the moment I get "Audio FAX not allowed on channel" |
| 18:36.29 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 18:37.06 | itsbroken | Hello, I'm looking for some advice as installing asterisk as a non-root from source. When i'm installing dahdi and libpri how can I specify where to install? DESTDIR? |
| 18:39.56 | *** join/#asterisk oej (~olle@95.209.191.71.bredband.tre.se) |
| 18:41.55 | p3nguin | Why can't you install as root? |
| 18:42.19 | itsbroken | I can but for security reasons I wanted to do a non-root install |
| 18:43.52 | *** join/#asterisk wonderworld (~ww@port-92-201-64-142.dynamic.qsc.de) |
| 18:46.44 | p3nguin | For dahdi, "make DESTDIR=/your/non-standard/non-root/path install" should work. |
| 18:46.54 | p3nguin | I don't use libpri, but I bet it's going to be the same. |
| 18:48.14 | p3nguin | For dahdi, don't forget "make DESTDIR=/your/non-standard/non-root/path modules" before you do the install. |
| 18:48.56 | p3nguin | Hope that helps. I've got to go get lunch before I don't have time to get lunch. |
| 18:52.16 | Charrit | Can I store in master.csv a channel variable? |
| 18:56.36 | torgnyw | Hi, we are trying to configure an Asterisk server as passthrough between ISP and an Alcatel PBX. Everything works well, except we can not dial out! As soon as we hit 0 (expected by Alcatel for outgoing calls), the call jumps to 's' extension in incoming context... Ideas? |
| 18:57.48 | torgnyw | Oh, we have 2 PRI cards... |
| 18:58.40 | mbowie | Which macro can I use to find my extension in a dialplan? CID_10000 has the CID we need, but ${EXTEN} isn't populated. :-/ |
| 19:05.43 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:05.53 | benngard | torgnyw: if i understand u corect, phone - alcatel - ei -asterisk - sip - pstn? |
| 19:09.46 | *** join/#asterisk f0ner00t (~f0ner00t@69-170-21-20.static-ip.telepacific.net) |
| 19:10.29 | f0ner00t | Hello. |
| 19:10.30 | torgnyw | benngard: phone -> Alcatel -> Asterisk(PRI card 1) <-> Asterisk(PRI card 2) -> PSTN. It's no VoIP involved, we just want to set the Asterisk server as transparrent as possible... |
| 19:11.42 | *** join/#asterisk imcdona (imcdona@pfsense.voicebyip.info) |
| 19:13.18 | benngard | torgnyw: and from the phone u dial for example 0 for external and then 12345 as callee? |
| 19:13.51 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 19:14.00 | torgnyw | benngard: Exactly. |
| 19:15.24 | benngard | then do somethimg like this in asterisk, just to test that u get the call: exten => 012345,1,Answer() |
| 19:15.45 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 19:15.53 | Aut0ExeC | anyone oppose running asterisk on a router? |
| 19:16.02 | Aut0ExeC | no fancy stuff... music on hold etcetc |
| 19:16.03 | benngard | if u se that call u know that the call is "in" your asterisk, and then u can remove thz zero |
| 19:18.29 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 19:18.30 | benngard | torgnyw: think u need core set verbose 3 to be able to se the call |
| 19:18.36 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 19:19.55 | Aut0ExeC | benngard: what u think about asterisk on a router? |
| 19:20.33 | torgnyw | benngard: Nothing really is transmitted from the phone to asterisk - as soon as you press 0, it goes to 's' in the context. If we have the following extensions.conf: exten => s,1,Playback(hello-world) we get the Hello world message as soon as we press 0 on the phone. |
| 19:21.39 | benngard | disbale over-lapped dialing? |
| 19:22.17 | torgnyw | benngard: where to I set this? |
| 19:22.46 | benngard | somewhere in tha alcatel, but w8, not a good idea |
| 19:24.11 | torgnyw | benngard: We really hope we can get this to work on the Asterisk side, as we don't have access to Alcatel config (or, we have to pay for it...). |
| 19:24.46 | benngard | on the phone, u press zero, the phone transmits the zero to asterisk, what if u make an extension like exten => 0,1,Answer(), does ithe call "land" there? |
| 19:28.36 | *** join/#asterisk vrtigo1 (~fredw@vpn.lpga.com) |
| 19:28.59 | torgnyw | benngard: still goes to 's'. Seems like hitting 0 just opens the channel, nothing else? |
| 19:29.55 | torgnyw | benngard: extension variable seems empty! |
| 19:30.09 | benngard | torgnyw: guessed so |
| 19:30.29 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:30.32 | benngard | i have an avaya, tinking og how that bastard works |
| 19:30.53 | benngard | attached to an asterisk |
| 19:33.53 | benngard | when u press 0, do u normally get a tone back, to indicate that u have "an external line"? |
| 19:35.03 | torgnyw | yes. |
| 19:36.14 | benngard | so u dial 0 get a tone and then the number (nomally i mean) |
| 19:36.26 | *** join/#asterisk Godofmonkeys (~Godofmonk@71-87-181-173.dhcp.jcsn.tn.charter.com) |
| 19:37.08 | torgnyw | Yes, all user must hit 0 first to get an external line. |
| 19:37.37 | torgnyw | ...when connected to Alcatel without asterisk between (normal usage). |
| 19:37.55 | benngard | thats what i ment |
| 19:38.34 | benngard | i am just guessing how the alcatel works now, so dont laugh at me plz, but we try somethig like this |
| 19:39.20 | benngard | exten s,1,Answer() |
| 19:39.36 | benngard | exten 1,1;Answer() |
| 19:39.45 | benngard | try dialing 0 1 |
| 19:39.52 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:40.34 | torgnyw | the ; is a typo right? |
| 19:40.55 | benngard | yes |
| 19:41.11 | benngard | sorry |
| 19:41.46 | p3nguin | itsbroken: Did that work out for you? |
| 19:42.36 | *** join/#asterisk reenignEesreveR (~sharjeel@116.71.185.210) |
| 19:43.38 | reenignEesreveR | is there a generic opensource billing system not specifically for asterisk but more or less any telco related thing? I am setting up an SMS gateway with kannel and i'd like to bill my users |
| 19:43.53 | Godofmonkeys | What is the reccommended front end, freepbx or Asterisk-gui? |
| 19:43.59 | Qwell | Godofmonkeys: vim |
| 19:44.17 | Godofmonkeys | indeed. |
| 19:44.28 | torgnyw | benngard: unfortunately, silence... |
| 19:45.01 | Godofmonkeys | so ideally i should not install any sort of gui? |
| 19:45.35 | benngard | torgnyw: silence is what i excpect but what do u se in asterisk with core set verbose 3 |
| 19:45.41 | p3nguin | godofmonkeys: vim is good enough. What else do you think you need? |
| 19:46.04 | Godofmonkeys | an interface for the non technical management types |
| 19:46.26 | p3nguin | Do you really want that type of person trying to admin your Asterisk system? |
| 19:46.36 | Qwell | non-technical management types shouldn't be mucking with the settings |
| 19:46.43 | torgnyw | benngard: s -> busy. we use verbose 6 right now. |
| 19:46.48 | Godofmonkeys | agreed. |
| 19:47.52 | Godofmonkeys | It has so far caused me more problems than its worth, but they seem to think it's a requirement |
| 19:48.04 | benngard | torgnyw: replace answer with hangup |
| 19:48.07 | p3nguin | It's not a requirement. |
| 19:48.07 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 19:48.25 | torgnyw | benngard: on both lines? |
| 19:48.35 | benngard | no just s extension |
| 19:49.11 | benngard | but i am just guessing now, need a dump to se whats really going on |
| 19:49.26 | torgnyw | right after s -> hangup. Doesn't bother with 1. |
| 19:49.50 | p3nguin | Are you having problems understanding dial plan logic or something? |
| 19:49.58 | benngard | exten s,1,Hangup() |
| 19:51.06 | benngard | exten => s,1,Hangup() |
| 19:51.57 | torgnyw | When we press 0 on the phone, we see Executing [s@contextname:1] Hangup("DAHDI/i1/22005521-13","") in new stack |
| 19:52.34 | torgnyw | Then Spawn extension (context, s, 1) exited non-zero on |
| 19:52.42 | benngard | s = start so we get the call |
| 19:53.39 | torgnyw | yes, the problem is that we are not able to press anymore keys on the phone, what can we put in extensions.conf to wait and listen for more digits from the phone |
| 19:54.08 | p3nguin | You could use Read() |
| 19:54.16 | p3nguin | But this really sounds like a PHONE problem. |
| 19:54.32 | p3nguin | You need to make it so the phone dials more numbers. |
| 19:54.39 | benngard | it is a phone/alcatel problem |
| 19:54.48 | torgnyw | How do we use Read()? Can we ask for a number of digits and then use this as extension to call |
| 19:54.57 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:54.59 | p3nguin | core show application Read |
| 19:55.10 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 19:55.23 | benngard | and if u put exten => s,1,Answer() what do u see |
| 19:55.27 | p3nguin | I'd be more interested in fixing the phones so you can dial an appropriate amount of digits, though. |
| 19:55.35 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 19:56.08 | *** join/#asterisk jkroon (~jkroon@dsl-241-240-52.telkomadsl.co.za) |
| 19:56.59 | torgnyw | Just to clearify, the Alcatel system is on the way out, Asterisk is a replacement. But we want to set it transparrent for a while when working on it. We have to pay to "fix" the Alcatel system, and we're tired of that. |
| 19:57.20 | torgnyw | benngard: answer, then busy |
| 20:02.23 | benngard | torgnyw: just add a Dial, after answer i mean, like exten => s,2,Dial(some number) |
| 20:03.27 | benngard | a sip extension or whatever |
| 20:03.39 | torgnyw | Then it calls my cellphone as expected. |
| 20:03.43 | p3nguin | Why do you want to Answer before Dial? That's not normal. |
| 20:04.15 | benngard | i know, i just want to se that we got the call |
| 20:05.36 | benngard | exten => s,1,NoOp(${EXTEN}) what does that give u? |
| 20:05.44 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 20:05.52 | chazzam | ~newbook |
| 20:05.53 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 20:06.30 | torgnyw | benngard: NoOp says 's' |
| 20:07.47 | benngard | NoOp(${CALLERID(dnind)}) |
| 20:09.13 | p3nguin | check spelling on that |
| 20:09.52 | benngard | see that, sorry, i should be to bed |
| 20:10.27 | torgnyw | so - wher's the typo? dnind? |
| 20:12.41 | *** join/#asterisk a_m_y (~simpleboy@112.204.229.14) |
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| 20:13.25 | p3nguin | I was thinking it was dnid. That's why I said check it rather than saying it was wrong. |
| 20:14.14 | torgnyw | empty. "" |
| 20:14.45 | benngard | ${CALLERID(DNID)} |
| 20:15.39 | benngard | but i think it will be empty :( |
| 20:16.56 | torgnyw | We beginn to fear this is not resolvable from the Asterisk side...? |
| 20:17.15 | benngard | last try before bed |
| 20:17.32 | benngard | exten => s,1,Answer() |
| 20:17.50 | benngard | exten => s,n,Wait(5) |
| 20:18.20 | benngard | exten => NoOp(${CALLERID(DNID)}) |
| 20:18.28 | benngard | add s,n |
| 20:18.45 | benngard | press 0 123 for example |
| 20:19.04 | p3nguin | If you'll use Verbose() instead of NoOp(), it's a lot easier to see what is printed. |
| 20:20.18 | benngard | u are ofc right, as i said, to late for me, my brain is more or less already on the pillow |
| 20:21.42 | torgnyw | benngard: thanks for your help. Unfortunately, Verbose is empty on the last try. |
| 20:22.00 | benngard | sorry |
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| 21:00.47 | mazpe | anyone recommends a decent trunk provider? |
| 21:02.00 | mazpe | actually all i'm looking for is outgoing calls. |
| 21:02.22 | sbrath | <PROTECTED> |
| 21:02.26 | sbrath | oops. |
| 21:02.40 | p3nguin | ~itsp |
| 21:02.40 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 21:02.43 | p3nguin | mazpe: ^^^^^^ |
| 21:02.59 | mazpe | ~itsplist-us |
| 21:02.59 | 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 |
| 21:03.24 | sbrath | I'm wondering if anyone has advice on why IAX2 is leaving like 50 channels around for a iaxmodem, and how I can get rid of them? |
| 21:04.22 | sbrath | should I use iax2 prune realtime all ? |
| 21:06.04 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 21:11.55 | *** join/#asterisk [shodan] (~shodan@dsl-67-55-6-124.acanac.net) |
| 21:14.07 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 21:16.19 | [shodan] | I just don't think that replacing ${TIMESTAMP} with ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) has improved user friendliness, configuration consistency, usability or reliability in any way ! |
| 21:16.55 | *** part/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 21:17.15 | [shodan] | in fact it sounds like this was done on purpose to make it harder to use asterisk, which doesn't make that much sense ! |
| 21:17.50 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 21:18.51 | p3nguin | Yeah, I bet that was the whole reason for the change: to make it harder to use Asterisk. |
| 21:20.22 | *** join/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com) |
| 21:21.00 | rgagnon | Q: does anyone have the singluar sound file 'hour", or is it just accidentally missing from the asterisk source? |
| 21:21.22 | rgagnon | (preferably in .ulaw format) |
| 21:23.39 | p3nguin | There's "hours," but no "hour." |
| 21:23.45 | rgagnon | correct |
| 21:24.00 | rgagnon | but there is minute, second,day, and year (as well as their plurals) |
| 21:24.40 | [shodan] | also SetVar SetAccount SetCallerID = ಠ_ಠ|
| 21:24.55 | *** join/#asterisk malconxx (~malconxx@unaffiliated/malconxx) |
| 21:25.01 | [shodan] | oh well |
| 21:25.17 | [shodan] | I suppose there was a 1.6 migration document that I never found |
| 21:25.51 | rgagnon | ... without an 'hour" prompt, it makes it hard to say something like "1 hour 5 minutes" |
| 21:26.15 | p3nguin | Can you really tell that much of a difference if it says 1 hours and 5 minutes? |
| 21:26.30 | rgagnon | yeah... since it won't sound professional |
| 21:27.25 | p3nguin | I can't remember which sound it was, but this is a familiar topic... we determined that it was impossible to discern the difference with and without the s. |
| 21:27.37 | [shodan] | attention to detail like that goes a long way rgagnon |
| 21:28.00 | [shodan] | you can probably clip the "s" sound using audacity |
| 21:28.43 | rgagnon | true. I was just seeing if maybe it was just missing and someone had allison saying that prompt already. maybe a small minor ticket to add it if I can get the file correct |
| 21:30.54 | p3nguin | It's not just missing... it doesn't exist in Asterisk nor Asterisk's sound packages. |
| 21:37.42 | itsbroken | p3n do I need to specify DESTDIR for the make all or just the make install? (does make all cover the make modules) |
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| 21:50.21 | Sertys | hrrr |
| 21:50.37 | Sertys | does SRTP have to be end-to-end |
| 21:50.46 | Sertys | afaik it's end-to-site |
| 21:51.22 | Sertys | so is there a problem bridging a client peer with srtp enabled and one that is plain sip/rtp |
| 21:51.54 | *** part/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com) |
| 22:05.55 | russellb | no, it is not end to end |
| 22:06.05 | russellb | it is decrypted (and potentially encrypted again) in asterisk |
| 22:06.13 | russellb | so you can, if you want, bridge srtp to a non-srtp call |
| 22:06.32 | russellb | if it meets your security needs ... |
| 22:07.41 | f0ner00t | Does anybody webmeetme 4? |
| 22:12.38 | Charrit | Can I store in master.csv a channel variable? |
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| 22:56.59 | Sertys | russellb: i am trying to deploy SRTP on nokia voip client, but client it doing whatever it wants |
| 22:57.41 | Sertys | is it necessary to set both a=crypto:n and RTP/SAVP for srtp to be considered enabled |
| 22:58.22 | Sertys | cuz client is sending a key in the a= param of the sdp request, but seems to be omitting the SAVP parameter in the audio request |
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