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| 07:23.48 | bof22 | Hi, I am new user and I looking for some help to develop Asterisk application with netbeans or eclipse, but I don't know how to do that. |
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| 07:25.08 | WIMPy | You need to elaborate on that. |
| 07:28.07 | bof22 | Ok, I need to develop an application like app_skel.c. I want to use Netbeans or Eclipse but when I import project with source (under Netbeans), the build is ok, but I can't debug app_skel.c with netbeans debugger. |
| 07:29.37 | WIMPy | Err, isnt' netbeans a Java thing? Or am I missing something. |
| 07:29.54 | WIMPy | But good to know that you meant "application" in the Asterisk sense. |
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| 07:32.12 | bof22 | Netbeans is also use to develop C or C++ application wtih the correct pluguin |
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| 10:14.30 | catphish | i'm just starting to use AGI, can it reject a call before it's ringing? |
| 10:17.35 | WIMPy | Just Hangup() |
| 10:19.06 | catphish | so at this point the UA has received a 100 in progress? can i then use Hangup() to throw it a 404? |
| 10:19.47 | WIMPy | Hangup(1) |
| 10:19.53 | catphish | WIMPy: fantastic, thanks |
| 10:21.33 | catphish | oh, one more question, is there a shortcut way to pass *all* calls to an AGI script (ie for all contexts) |
| 10:21.57 | WIMPy | No. |
| 10:21.59 | catphish | (otherwise everyone will need a dialplan entry, which isn't a huge problem) |
| 10:22.05 | catphish | thanks :) |
| 10:22.18 | WIMPy | You can't do anything for multiple contexts. That would somehow defeat the purpose anyway. |
| 10:22.41 | catphish | i was assuming the context was passed to the AGI script |
| 10:22.54 | WIMPy | If you want all calls to go to an AGI, make sure they go to the same context., |
| 10:23.15 | catphish | maybe that's the answer, as long as there's some other way AGI can reliably identify the source |
| 10:23.16 | WIMPy | You can read it, sure. |
| 10:23.39 | catphish | i'll look through the available variables |
| 10:23.55 | catphish | "agi_channel" should do the job |
| 10:25.53 | catphish | taking a step back, is this a sane thing to be doing (assuming i want to control my whole dialplan with AGI) |
| 10:26.10 | catphish | (putting all untrusted SIP end users in the same context that is) |
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| 10:40.04 | catphish | think i prefer using contexts to identify things :) |
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| 11:00.08 | catphish | works great, thanks for the pointers WIMPy :) |
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| 12:24.29 | Czindy | Hello, I would like to ask for your help, how can I delay asterisk start after MariaDB on a fresh Centos7 - Asterisk 13 - MariaDB 5.5 install. The issue is that on server reboot, asterisk behavior is strange. The ODBC shows no connections and all the realtime / cdr backends are not working. |
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| 12:56.31 | Phil-Work | is it possible to make multiple registrations against a single SIP server (different username/password for each)? |
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| 12:58.29 | flujan | Hello guys. I am trying to detect tones/digits after a call is answered by the user. I know that asterisk keeps monitoring the channel for transfer purposes. Like the user pressed the * or # key. |
| 12:58.51 | flujan | So, how can I log it on the consoleif the key is not mapped on the features.c |
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| 13:13.20 | carrar | Phil-Work, thats typically how a office full of phones behind a router running NAT works |
| 13:14.19 | carrar | flujan: try a higher debug mode |
| 13:14.31 | carrar | or perhaps log high level debug to a log file |
| 13:14.47 | Phil-Work | carrar, I mean registering Asterisk against a single sip server multiple times |
| 13:14.55 | Phil-Work | rather than multiple phones registering against Asterisk |
| 13:17.40 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:18.17 | WIMPy | flujan: Depends. Asterisk will only monitor for DTMF input if any features are enabled on that channel. |
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| 13:20.25 | [TK]D-Fender | Phil-Work: Yes |
| 13:22.20 | Phil-Work | [TK]D-Fender, does the username part of the register just need to match the defaultuser of the relevant block in sip.conf? |
| 13:22.50 | [TK]D-Fender | It has absolutely nothing to do with a section definition. |
| 13:23.54 | Phil-Work | [TK]D-Fender, so when a call comes in, how does it determine the context it should be routed to? |
| 13:24.10 | Phil-Work | by the "host" of the section? |
| 13:24.29 | [TK]D-Fender | Depends on what kind of entry you make |
| 13:24.35 | [TK]D-Fender | And what the call looks like |
| 13:25.28 | [TK]D-Fender | Do they send a username? Do they send calls with auth at all? |
| 13:25.54 | Phil-Work | they do have a username |
| 13:25.55 | Phil-Work | no auth |
| 13:27.15 | Phil-Work | though the username is derived from the contact header sent in the registration |
| 13:27.29 | [TK]D-Fender | When they SEND you calls? |
| 13:28.09 | Phil-Work | yes |
| 13:28.31 | Phil-Work | the calls are sent to username@IP where username is the username taken from the contact header Asterisk sent in the registration |
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| 13:31.00 | [TK]D-Fender | That is a TO, not a FROM |
| 13:31.52 | Phil-Work | indeed |
| 13:32.17 | Phil-Work | the FROM is going to be the CLI of the original caller - that could be anything |
| 13:33.00 | WIMPy | That's the way things usually work. |
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| 13:35.26 | [TK]D-Fender | Which basically sounds like every other provider... who sends un-authed calls an not FROM to match them. |
| 13:35.44 | [TK]D-Fender | Therefor "type=" will be ... ? |
| 13:36.46 | bof22 | I need to develop an application like app_skel.c. I want to use Netbeans or Eclipse but when I import project with source (under Netbeans), the build is ok, but I can't debug app_skel.c with netbeans debugger. |
| 13:37.07 | WIMPy | Well, actually not. I usually see the CallerID in the From, weheras the CLI goes in to PAI. |
| 13:37.25 | bof22 | There is someone who meet this issu ? |
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| 15:31.03 | eschmidbauer | <PROTECTED> |
| 15:31.27 | eschmidbauer | <PROTECTED> |
| 15:32.00 | igcewieling | eschmidbauer: Asterisk will ignore the IPs in the SDP if correctly configured. |
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| 15:32.18 | eschmidbauer | how does it know where to route the media? |
| 15:32.38 | eschmidbauer | it will use the public IP + RTP Port? |
| 15:33.06 | igcewieling | I think it uses the source of the incoming RTP packets. |
| 15:33.14 | igcewieling | What are you trying to accomlish? |
| 15:33.23 | igcewieling | There are lots of documentation on NAT with Astersik. |
| 15:33.35 | eschmidbauer | we have intermittent one-way issues |
| 15:33.42 | ryanxm | is it possible to use a provider that needs a special dial pattern as a backup to one that doesnt? vitelity is shitting itself the past few days and looking into some alternatives |
| 15:33.54 | eschmidbauer | i'm 80% sure it's the cisco router at fault |
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| 15:34.47 | igcewieling | From sip.conf.sample nat = comedia ; Send media to the port Asterisk received it from regardless ; of where the SDP says to send it. |
| 15:35.19 | igcewieling | eschmidbauer: disable SIP ALG on your Cisco router and configure Asterisk for NAT |
| 15:35.58 | ryanxm | that or does anyone know any good backups to Vitelity? |
| 15:36.08 | igcewieling | ~itsp |
| 15:36.08 | 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. |
| 15:36.46 | igcewieling | ryanxm: yes, you can configure Asterisk to route your calls in any way you want. |
| 15:36.53 | eschmidbauer | igcewieling: already done |
| 15:37.20 | igcewieling | eschmidbauer: then the issue is not your Cisco router. Chances are you messed up the Asterisk NAT configuration |
| 15:37.29 | eschmidbauer | but it's intermittent |
| 15:37.44 | eschmidbauer | we have nat = yes |
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| 15:37.48 | eschmidbauer | in sip.conf |
| 15:37.55 | igcewieling | eschmidbauer: that is a very old setting. |
| 15:38.01 | eschmidbauer | we are on asterisk 1.8 |
| 15:38.57 | igcewieling | Exactly how is NAT involved? Asterisk and phones behind same nat and service provider on the internet? Asterisk behind nat and phones out on the internet? |
| 15:39.27 | igcewieling | the answers to those questions will tell us how to configure nat for your needs. |
| 15:40.13 | ryanxm | igcewieling: so it is possible to use provider1 which needs no modifications to extensions.conf but if we fall back to provder2 which does require some kind of prefix it will still go though? --sorry new to this stuff it was just kinda thrown at me |
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| 15:40.31 | igcewieling | huh? no, you need to set up your dialplan to call the other provider when the first one fails. |
| 15:40.45 | igcewieling | See HANGUPCAUSE and DIALSTATUS channel variables. |
| 15:41.08 | ryanxm | ah ok that makes sense, and there I can still specify the needed prefixes that the second provider will need |
| 15:41.11 | igcewieling | ryanxm: sounds like you are running FreePBX. |
| 15:41.11 | ryanxm | thank igcewieling |
| 15:41.37 | ryanxm | freepbx w/ asterisk backend yeah |
| 15:41.51 | igcewieling | ryanxm: The answers I provided assume Asterisk, not FreePBX |
| 15:42.02 | ryanxm | i can jump over to #freepbx |
| 15:42.11 | igcewieling | To do this in FreePBX you should ask on #FreePBX |
| 15:42.13 | ryanxm | yeah no worries, figured if it would work with asterisk there is a chage |
| 15:42.27 | ryanxm | thank |
| 15:42.28 | ryanxm | s |
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| 15:50.51 | eschmidbauer | igcewieling: phones are behind NAT |
| 15:50.55 | eschmidbauer | asterisk is on public |
| 15:51.18 | eschmidbauer | igcewieling: we have 1000s of phones *not* having this issue |
| 15:51.23 | eschmidbauer | it's only a few |
| 15:51.28 | eschmidbauer | so i dont think it's asterisk |
| 15:51.36 | eschmidbauer | but i haven't ruled it out tyet |
| 15:51.47 | eschmidbauer | i really think it's these cheap cisco 867s |
| 15:52.46 | [TK]D-Fender | Why do you think that? |
| 15:53.12 | [TK]D-Fender | You should know what side is acting wrong. That's what SIP debug is for. |
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| 15:59.10 | eschmidbauer | it's SDP |
| 15:59.30 | eschmidbauer | i have SIP debug enabled |
| 15:59.42 | eschmidbauer | or RTP rather |
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| 16:02.09 | igcewieling | eschmidbauer: routers often timout udp nat translations after 60s of no activity. Increase that on your router or set qualifyfreq to less than 60 |
| 16:03.09 | igcewieling | and stop obsessing on the SDP. If you have NAT configured on Asterisk the IP in the SDP doesn't matter and is ignored. |
| 16:04.51 | eschmidbauer | igcewieling: already increased NAT timeout |
| 16:05.10 | eschmidbauer | okay. but this is a media issue. |
| 16:05.46 | eschmidbauer | anyway, we rebooted the cisco and problem seems to be resolved |
| 16:07.10 | igcewieling | best of luck. |
| 16:08.39 | eschmidbauer | thanks |
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| 19:24.00 | Echo6 | I cannot seem to get my damn PBX to stop sending voicemail as asterisk@ |
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| 19:44.08 | pjensen00 | Hello, I'm wondering if anyone knows how to match an extension with a leading underscore? Currently asterisk is generating Local/_attended@transfer channels on a REFER that as a "Replaces" portion of the Refer-To header. |
| 19:44.15 | pjensen00 | And the only way I can match it is with _. |
| 19:44.23 | pjensen00 | Which of course leads to a whole host of new issues. :( |
| 19:45.02 | robmal | \_ ? |
| 19:45.07 | pjensen00 | doesn't seem to work. :( |
| 19:45.46 | robmal | So why do you need _ in extension anyway? |
| 19:46.31 | pjensen00 | Asterisk is creating it, I don't have a choice. |
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| 19:46.45 | robmal | Asterisk isn't creating anything by itself. |
| 19:47.05 | pjensen00 | Asterisk is generating Local/_attended@transfer channels on a REFER that as a "Replaces" portion of the Refer-To header. |
| 19:47.09 | [TK]D-Fender | put it in your pattern |
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| 19:47.15 | pjensen00 | Tried that. |
| 19:47.23 | [TK]D-Fender | tried wrong.... |
| 19:48.23 | pjensen00 | \_attended, _[_attended], _[\_attended], _attended!, none work. |
| 19:48.43 | pjensen00 | __attended or _attended either. |
| 19:49.49 | robmal | My asterisk is generating Local/$ext-$n+1;1 and Local/$ext-$n+1;2 ;-> |
| 19:50.07 | pjensen00 | Sorry, I did leave out one important part. The refering channel is in Stasis. |
| 19:50.44 | [TK]D-Fender | Indeed all of those are wrong |
| 19:50.53 | pjensen00 | Which is why I have asked for help. |
| 19:51.04 | pjensen00 | Otherwise, I would not have asked. |
| 19:51.19 | [TK]D-Fender | failed your basics |
| 19:51.23 | pjensen00 | This is unfortunate. |
| 19:51.29 | [TK]D-Fender | _attended, _[_attended], _[\_attended] <- everything in [] is SINGLE CHAR MATCH |
| 19:51.55 | [TK]D-Fender | [0-9] will match a single digit in that range |
| 19:52.05 | [TK]D-Fender | [12345] will match any ONE o fthose digits |
| 19:52.12 | [TK]D-Fender | Now think about your pattern again |
| 19:57.13 | [TK]D-Fender | pjensen00__attended <--- you've also forgotten that on its own "n" is a PATTERN CHARACTER |
| 19:57.45 | [TK]D-Fender | It'll break that down one char at a time.... and as-written, will fail |
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| 20:16.57 | moe` | hey [TK]D-Fender, how's it going dude? |
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| 22:56.28 | igcewieling | pjensen00: _[_]attended might do it. |
| 22:56.40 | pjensen00 | already tried it. :( |
| 22:56.50 | igcewieling | hmm. should not even need the leading underscore. |
| 22:57.19 | pjensen00 | Anyways, I think that weird transfer channel only shows up on weird looking REFERs. |
| 22:57.48 | pjensen00 | There's a bug with our SIP B2BUA that is sending blind transfers as Attended. Aka adding the "Replaces" portion in the Refer-To header |
| 22:58.39 | pjensen00 | So if the Replaces points to the same dialogue that sent it, Asterisk ends up crashing. |
| 22:59.08 | WIMPy | sip or pjsip? |
| 22:59.12 | igcewieling | This is from a production system: exten => _co[n]fbridge^1NXXNXXXXXX,1,Noop(Inter-node Conference Trunk Request) |
| 22:59.28 | pjensen00 | Though a series of events of course. The killer being asterisk trying to bridge the local channel into a stasis app when it can't extract an application name. |
| 22:59.40 | igcewieling | so we know it can be done with other characters. |
| 22:59.43 | pjensen00 | Which seems to be the last thing asterisk does before it explodes |
| 22:59.55 | pjensen00 | WIMPy: pjsip |
| 23:00.11 | pjensen00 | and it ONLY happens in refers to channels in a Stasis app |
| 23:03.40 | pjensen00 | igcewieling: that _co[n] extension, does the actual channel have the leading underscore? |
| 23:04.58 | igcewieling | pjensen00: it does not, I was only showing it *can* be done with other characters. perhaps it would be easy to add _ to the list of chars in the code? |
| 23:06.03 | pjensen00 | Ah I see. I don't think I'll actually need to detect in anyhow. The crashy bit was caused not by me not being able to match that pattern. |
| 23:06.20 | pjensen00 | It just so happend to happen at nearly the exact moment in time. :( |
| 23:06.56 | pjensen00 | The awesome part is that if I just don't DO anything and let a good REFER do its automatic bits, it just works. |
| 23:07.18 | pjensen00 | Which makes me a happy little Ithorian |
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| 23:19.52 | carrar | Nippon Online! |
| 23:20.38 | pjensen00 | I'm only half. :( |
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| 23:45.03 | [TK]D-Fender | igcewieling |
| 23:45.03 | [TK]D-Fender | pjensen00: _[_]attended might do it. <- nope. |
| 23:45.48 | pjensen00 | Correct. |
| 23:46.48 | [TK]D-Fender | becasue of poor escaping |
| 23:55.19 | rmudgett | _[_]atte[n]ded |
| 23:56.52 | WIMPy | Yes, the need to escape soem rather normal characters is extremely annoying. I also spent a lot of time with that one once. |
| 23:57.14 | WIMPy | Especially as I thought up to then that only the upper case letters would work as wildcards. |
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| 23:58.43 | WIMPy | I'm not even sure this is documented anywhere. |