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| 05:42.24 | tbsky | hi! my ISP need to use g.711 with VAD/CNG. so I need a sip server/proxy between my ISP and my asterisk to do transcoding. any suggesion for this kind of software? can rtpproxy or Kamailio or freeswitch do this kind of work? |
| 05:42.42 | tbsky | hope I can install the software at the existing asterisk box.. |
| 05:43.47 | tbsky | I think neither sip nor pjsip in asterisk can handle g711 with VAD? |
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| 06:26.11 | abyz | Hi everyone! |
| 06:26.28 | abyz | I'm a noob here so please bear with my questions |
| 06:27.20 | abyz | I'm Trying to terminate EFT-POS HDLC (V21, V23) using DAHDI or asterisk, but until now I don't have any success |
| 06:27.47 | abyz | my first solution the probelm was to define iaxmodem to have a modem interface |
| 06:28.40 | abyz | and then try to terminate HDLC connection there. I've tried to figure out how efax and hylafx work for this manner |
| 06:28.56 | abyz | but couldn't find the answer |
| 06:29.38 | abyz | after searching a little more I found out that maybe dahdi can handle HDLC and give me the frames |
| 06:30.00 | WIMPy | No |
| 06:30.17 | WIMPy | You first have to demodulate the audio to get data. |
| 06:30.44 | WIMPy | DAHDI can handle HDLC if it's transmitted digitally. |
| 06:30.45 | abyz | and I also have read dahdi and dahdi-tools README and documentation about HDLC in system.conf |
| 06:31.02 | abyz | so I need to use something like spandsp |
| 06:31.03 | abyz | for that |
| 06:31.12 | WIMPy | yes |
| 06:31.30 | abyz | and then try to extract hdlc frame |
| 06:31.41 | abyz | so using iaxmodem aproach seems more logical |
| 06:31.48 | WIMPy | yes |
| 06:32.10 | WIMPy | It's the most obvious one. |
| 06:32.30 | abyz | WIMPy, I told you that I'm noob ;) |
| 06:33.35 | abyz | WIMPy, Thanks. You made my mind clear |
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| 06:52.36 | qloogkm | anybody know if there is a way to make asterisk (either chan sip or pjsip) send the 'stale=true' parameter when a phone re-registers with an old nonce? |
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| 06:53.41 | qloogkm | chan_sip with asterisk 13 sends a new nonce but doesn't include this message saying the old one was stale |
| 06:54.04 | qloogkm | not a problem, but i wondered if we could set it to do that |
| 06:55.46 | WIMPy | You've got the source! |
| 06:57.33 | qloogkm | Fair answer, I don't speak C very well, that's why I asked :) |
| 06:58.15 | WIMPy | Might still be easy to add, even if you're not that good at C. |
| 07:00.29 | qloogkm | grep time |
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| 07:10.22 | qloogkm | pjsip sends it out by default, super, solved |
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| 08:35.46 | bulkorok | somebody know opus able hardphones? |
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| 09:56.36 | vassilux | I installed dahdi-source debian package but dahdi_scan can see my carte but dahdi_harwdare identify well the carte? Mya be I'm missing something from the installation process , Is there a documenation for this one ? |
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| 10:08.21 | WIMPy | dahdi_hardware tells you what drivers should be suitable for the detected cards. |
| 10:09.04 | WIMPy | dahdi_scan will show you which ones are atually available. |
| 10:13.30 | vassilux | WIMPy : yes dahdi_hardaware show me wcte43x+ |
| 10:14.18 | vassilux | WIMPy : lsmod | grep dahdi -> wcte43x is loaded |
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| 10:14.59 | WIMPy | The + also tells you that the driver is loaded. |
| 10:17.46 | vassilux | I had the same problems few days ago ... and find the solution but not remeber , I installed dahdi-complete from source |
| 10:18.28 | WIMPy | Have you configured your channels? |
| 10:25.03 | vassilux | non |
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| 10:47.09 | madduck | i have verbosity and debug levels set to 1000 in rasterisk, and Verbose() output is shown, but nothing else. How can I enable dialplan traces? I am used to seeing those, no idea why they don't show up. |
| 10:47.15 | madduck | v11.13 |
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| 11:26.51 | madduck | so I have _X. defined in a context, and an included context defines _77.. Shouldn't the latter take precedence when a 77* number is called? |
| 11:27.31 | WIMPy | no |
| 11:27.53 | WIMPy | Included contexts are only searched when there's no match. |
| 11:31.17 | madduck | :/ |
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| 11:55.48 | madduck | 'sip show peer foobar' includes Variables that I set, specifically INTL_CC = 49 |
| 11:56.15 | madduck | however, in the associated context, those variables are not available. Any idea? |
| 11:57.09 | madduck | it does work in other cases where a SIP phone is logged in, so that when a call originates in a context, the variables are available |
| 11:57.39 | madduck | the peer/context in question relates to a SIP registry, i.e. sipgate |
| 11:58.24 | madduck | could it be that "register =>" actually bypasses the whole sip stanza? |
| 11:59.12 | madduck | how else would it know which context to start in though? |
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| 12:03.38 | carrar | . |
| 12:03.40 | carrar | err |
| 12:03.45 | carrar | moof |
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| 12:11.02 | carrar | *sigh* https://libreplanet.org/wiki/Save_WiFi/Individual_Comments |
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| 12:14.57 | [TK]D-Fender | madduck: .. |
| 12:15.01 | [TK]D-Fender | ~siprgister |
| 12:15.04 | [TK]D-Fender | ~sipregister |
| 12:15.04 | infobot | [~sipregister] SIP registration is to tell your provider what IP address & EXTEN to send INCOMING calls to. Some ITSPs let you use a fixed address or host rather than registering. Registration is NOT normally needed to PLACE calls, as those are typically auth'ed independently. Others accept unauth'ed calls once you are registered (saves on negotiation BW). |
| 12:16.13 | [TK]D-Fender | madduck |
| 12:16.15 | [TK]D-Fender | so I have _X. defined in a context, and an included context defines _77.. Shouldn't the latter take precedence when a 77* number is called <- This is well documented. the actual context is searched. If there is no match then it goes on to looking in each included context in order |
| 12:17.58 | madduck | [TK]D-Fender: yes, I know what sipregister is, and that the way it cooperates with a sip.conf stanza is a bit weird and hackish. But the call comes in and gets routed to the 'trunk-sipgate' context in the dialplan, so there *must* be some sort of interaction with the sip stanza. Just the variables don't get set |
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| 12:18.06 | madduck | I just sent a mail about this to the list |
| 12:18.14 | [TK]D-Fender | It DOESN"T cooperate with them |
| 12:18.17 | [TK]D-Fender | they are independent |
| 12:18.31 | madduck | how does it know which context to enter the dialplan in? |
| 12:18.31 | [TK]D-Fender | It will match whatever that call matches when it comes in. |
| 12:18.48 | [TK]D-Fender | Register tells them where to send the call. It has nothing to do with matching |
| 12:19.05 | [TK]D-Fender | And what is this about variables being set? |
| 12:19.12 | madduck | setvar |
| 12:19.15 | [TK]D-Fender | Show us |
| 12:19.24 | [TK]D-Fender | the config and the call |
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| 12:19.51 | madduck | http://slexy.org/raw/s21NsfDKyW |
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| 12:20.51 | madduck | so right. sipgate sends a call to 'incoming', and sip.conf matches that according to type=peer and host=XXX so that the call is associated with SIP entity 'trunk-sipgate' when it arrives |
| 12:21.12 | madduck | and when it arrives, it enters the dialplan accordingly at in-trunk-sipgate |
| 12:22.01 | madduck | which is analogous to how it works when my type=friend,host=dynamic SIP phone originates a call |
| 12:22.19 | [TK]D-Fender | Setvar works fine |
| 12:22.33 | [TK]D-Fender | show us the dialplan as well and the call |
| 12:23.37 | madduck | what does it mean to show "the call" |
| 12:23.38 | madduck | ? |
| 12:24.00 | madduck | here is the relevant bits from the dialplan: http://slexy.org/raw/s21B1vEdFs |
| 12:24.01 | [TK]D-Fender | Actually LOOK at the call at * CLI with SIP debug enabled\ |
| 12:24.14 | madduck | invalidHandler just defines a standard i handler |
| 12:24.59 | [TK]D-Fender | context=in-trunk-sipgate |
| 12:25.08 | [TK]D-Fender | [lehel-in-trunk-sipgate] |
| 12:25.13 | [TK]D-Fender | NOT the same context at all |
| 12:25.17 | madduck | right |
| 12:25.22 | madduck | i edited the message |
| 12:25.29 | [TK]D-Fender | Why are you showing mismatched items? |
| 12:25.44 | [TK]D-Fender | My trust in what you're showing is dropping. |
| 12:25.58 | [TK]D-Fender | Show proper things that are supposed to match and go get a call to show us |
| 12:26.01 | madduck | context=lehel-in-trunk-sipgate |
| 12:26.11 | madduck | is in the sip.conf stanza |
| 12:26.37 | madduck | i just don't want to put too much info into the public, that is why |
| 12:29.04 | _fuz | Hi. I wonder about asterisk configuration files. Is it parsing any file which ends with .conf in /etc/asterisk/ or is it just looking for predefined files like extensions.conf and is ignoring something like extensions-XYZ.conf? Thank you. |
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| 12:30.54 | [TK]D-Fender | _fuz: always predefined |
| 12:31.14 | [TK]D-Fender | _fuz: Otherwise it'd have no idea what it was a config FOR |
| 12:33.13 | _fuz | Thank you. I am searching errors in an existing installation and found backup files ending with .conf and I learned about not to use that name i.e. for apache where all files with extension .conf are parsed, no matter what name. |
| 12:35.43 | _fuz | I have an old 1.2.16 productive environment which does not take incoming calls anymore suddenly after it runs for hours. Restart and it works again... |
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| 12:37.16 | [TK]D-Fender | That is a seriously decrepit system..... |
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| 12:38.11 | _fuz | Yes, but there is no new one set up and 50 agents need it go get things going. |
| 12:40.27 | [TK]D-Fender | _fuz: Lots of things could come into play for this. Could a networking issue witha timeout where VoIP is involved, could just be a bug causing an eventual lock-up. |
| 12:42.45 | _fuz | Yes, it gives me headaches. |
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| 12:50.13 | madduck | [TK]D-Fender: I hope this is now a complete and suitable test case: http://slexy.org/raw/s21uib1U1D |
| 12:51.55 | [TK]D-Fender | madduck: Doesn't match |
| 12:52.21 | [TK]D-Fender | madduck: [Sep 2 14:41:53] Found peer 'telco-trunk-sipgate' for '017xxxxxxxx' from 217.10.79.9:5060 |
| 12:52.36 | madduck | wtf? |
| 12:52.37 | [TK]D-Fender | madduck: Clearly not a context you've shown us |
| 12:52.48 | madduck | OH |
| 12:53.05 | madduck | so there are two separate registrations for sipgate |
| 12:53.15 | madduck | and I have no way to tell them apart, right? |
| 12:53.23 | madduck | other than the extension they send |
| 12:53.24 | [TK]D-Fender | it hit THAT one. |
| 12:53.36 | madduck | right, this now makes a lot of sense |
| 12:53.36 | [TK]D-Fender | because it matched first |
| 12:53.38 | madduck | thanks for your patience |
| 12:53.56 | madduck | this sip stanza matching is a bit unfortunate for my use case |
| 12:57.13 | madduck | [TK]D-Fender: thank you! |
| 12:57.27 | [TK]D-Fender | madduck: You're welcome. |
| 12:59.29 | madduck | it now works |
| 12:59.39 | madduck | hackishly but i see no better way |
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| 13:01.36 | vassilux | For my dahdi_scan problem, create /etc/init.d/dahdi file and restart dahdi gives me a solution |
| 13:08.57 | _fuz | I found this in the logs: ERROR chan_sip.c: We could not get the channel lock for SIP/... SIP message just ignored: bye ... BAD! BAD! BAD! Could this be a serious thing causing to not route incoming calls anymore? |
| 13:09.21 | [TK]D-Fender | Maybe. |
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| 13:14.58 | tnt | I'm having an issue when trying to apply a 'userfield' to the CDR records in case of call being transferred. If I have a call coming from A (IAX) -> B (SIP) and then B does an SIP attended transfer to C (SIP), I get two records A->B and B->C but the 'userfield' that I set initially in the dial plan on all calls coming from A ends up in the B->C record instead of the A->B one ... |
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| 13:54.04 | malinens | <PROTECTED> |
| 14:09.26 | [TK]D-Fender | there is no such thing as "all calls" |
| 14:09.31 | [TK]D-Fender | there are only the patterns you make |
| 14:09.45 | [TK]D-Fender | next, why would you have a pattern there and then be dialing the same device all the time? |
| 14:10.05 | [TK]D-Fender | Also, every call is "incoming". |
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| 14:34.24 | phpboy | malinens: what is the alternative to dialing 21? I imagine you want to dial the extention being dialed which can be done by replacing SIP/21 with SIP/${EXTEN} |
| 14:41.59 | [TK]D-Fender | That will use whatever was dialed. |
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| 14:46.06 | phpboy | bingo |
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| 14:59.07 | _fuz | [TK]D-Fender: I also do have a lot of warnings of type: 'channel.c: Avoided deadlog for xyz 10 entries' in the log. I am not sure if I should track this down. Do you know what deadlock is meant? |
| 14:59.31 | _fuz | retries, not entries |
| 15:00.06 | [TK]D-Fender | channel locking issue... which I suppose might cause the entire channel driver to back up. |
| 15:01.27 | _fuz | to back up? Sorry what do you mean by "back up"? |
| 15:04.58 | jdean141 | Hi. I try to call via dahdi (using Dial(DAHDI/1)) and I get the following error: Channel 0/1 got hangup, cause 27. Any ideas what causes this? |
| 15:07.57 | _fuz | isdn cause 27 is defined by "Destination out of order". |
| 15:09.48 | jdean141 | _fuz, I have E1 trunk and try to go through it. The trunk itself seems to be working, if I write to /dev/dahdi/.. I can read it in the other side. So why does asterisk think it's out of order? |
| 15:13.10 | _fuz | sorry, dunno |
| 15:16.20 | [TK]D-Fender | Show us the actual call. |
| 15:16.40 | [TK]D-Fender | _fuz: "back up" -> seize. Lock. Freeze. Block. Take your pick |
| 15:17.44 | [TK]D-Fender | ~pb |
| 15:17.44 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:17.45 | [TK]D-Fender | ^^^ |
| 15:19.26 | jdean141 | http://pastebin.com/DZmPpwjF |
| 15:19.49 | [TK]D-Fender | you arewn't even dialing a number |
| 15:20.06 | [TK]D-Fender | of course it's going to reject you |
| 15:20.07 | jdean141 | you mean DAHDI/1/2001 for instance? It doesn't work, same problem |
| 15:20.25 | [TK]D-Fender | Show us the new attempt |
| 15:20.33 | [TK]D-Fender | and have PRI DEBUG enabled |
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| 15:26.25 | jdean141 | http://pastebin.com/JdqVDS7g |
| 15:26.35 | [TK]D-Fender | full |
| 15:26.36 | [TK]D-Fender | attempt |
| 15:26.41 | [TK]D-Fender | not just little snippits |
| 15:27.31 | jdean141 | It's problematic, I can't really copy it. The asterisk is not connected to the internet, so I copy by hand |
| 15:27.37 | jdean141 | : |
| 15:27.41 | jdean141 | :( |
| 15:29.08 | _fuz | ssh to the asterisk and copy using x? |
| 15:29.47 | jdean141 | It is in a different network, I cannot connect it to the internet |
| 15:30.02 | jdean141 | let me know what to look for in the logs? |
| 15:38.53 | [TK]D-Fender | We can't tell you what to lok for since we don't know what's wrong and what's actually getting passed |
| 15:39.02 | [TK]D-Fender | this is a very bad guessing game |
| 15:40.44 | jdean141 | Ok, I'll copy by hand. This will take a while |
| 15:42.14 | [TK]D-Fender | Are you stuck physically at the machine and that's it? |
| 15:44.27 | jdean141 | there's a local lan |
| 15:44.57 | [TK]D-Fender | And NO internet access from the LAN? |
| 15:45.43 | jdean141 | that's correct |
| 15:50.08 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 15:51.26 | [TK]D-Fender | I'd see about connecting whatever you're typing to us on right now to it if you are physically there... |
| 15:51.45 | [TK]D-Fender | or grabbing pics off a phone... or whatever |
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| 16:06.23 | jdean141 | http://pastebin.com/RQVYA4eP |
| 16:08.12 | jdean141 | http://pastebin.com/Eh5GCwxX - fixed a slight copying mistake in the first line |
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| 16:10.35 | [TK]D-Fender | == Primary D-Channel on span 1 up |
| 16:10.45 | [TK]D-Fender | I'm getting the impression of a bouncing D-Channel |
| 16:13.59 | jdean141 | So what should I do? |
| 16:14.57 | [TK]D-Fender | machine isn't really connected, you're running absolutely ancient garbage and hopig to "debug". This is a massive exercise in pain. |
| 16:15.09 | [TK]D-Fender | What kind of deployment is this? |
| 16:15.26 | [TK]D-Fender | What kind of stuff is it doing? |
| 16:19.28 | jdean141 | Currently I have a single computer with asterisk and a digium. the digium is connected to itself (loopback) for testing that it works |
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| 16:25.13 | WIMPy | jdean141: It does not look like the interface you're trying to use is connected to something usable. |
| 16:27.32 | WIMPy | And you don't want to dial on a cahnnel anyway. Use a group. |
| 16:27.36 | jdean141 | As I said, I connected it to itself (loopback) for checking that it works. Is there any reason why this wouldn't work? |
| 16:28.01 | WIMPy | What kind of loopback? |
| 16:28.12 | jdean141 | e1 loopback cable |
| 16:28.13 | Ice_Strike | Is it possible to migrate agents and active call waiting from a Queue to another Queue on different Server? |
| 16:28.18 | WIMPy | To the next port? To the same port? |
| 16:28.27 | jdean141 | to the next port |
| 16:29.01 | WIMPy | Then that next port isn't configured for that loop. |
| 16:29.13 | WIMPy | Ice_Strike: Need more input. |
| 16:29.36 | jdean141 | What do you mean by not configured? |
| 16:30.15 | WIMPy | It's not acting as a peer to the one you're uding. |
| 16:30.22 | Ice_Strike | let say 1 big queue and 10 small ones, is it possible dynamically move small ones to another server to give more resources to big one? |
| 16:31.15 | WIMPy | So you want it automatically? Well, you can just 'redirect' all channels. Not sure how practical it would be. |
| 16:31.36 | [TK]D-Fender | Ice_Strike: there is no such thing as "migrate" |
| 16:32.10 | [TK]D-Fender | you can violently toss them out of the queue and try to redirect the entire call off where you'd have appropriate dialplan set up to accept them and dump them in the right place. |
| 16:32.15 | [TK]D-Fender | Nice & messy |
| 16:33.55 | jdean141 | WIMPy, so if I understand correctly, if I'll connect it to another interface in another computer it should be fine? |
| 16:34.34 | WIMPy | It doesn't matter where it is physically. It has to be configured appropriately. |
| 16:35.24 | jdean141 | inside asterisk or outside of it? (where do I look for what I should configure) |
| 16:35.34 | WIMPy | Both. |
| 16:35.37 | [TK]D-Fender | WIMPy: He's on ----> 1.2 <---- |
| 16:35.51 | WIMPy | First DAHDI needs a config and then Asterisk. |
| 16:36.10 | WIMPy | 1.2 doesn't know DAHDI. |
| 16:36.38 | Ice_Strike | [TK]D-Fender Ah I see.. I have a idea that all queues should be in 1 DB and external algorithm will assign server id to queue based on queue members and resource metrics :) |
| 16:36.46 | WIMPy | Are you thinking about _fuz? |
| 16:37.00 | [TK]D-Fender | Aim might be off... |
| 16:37.24 | [TK]D-Fender | indeed |
| 16:37.26 | [TK]D-Fender | strike that |
| 16:39.59 | jdean141 | alright, I'll upgrade asterisk and try to reconfigure. thanks |
| 16:43.12 | _fuz | Anyone talking to me - just woke up :-) |
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| 17:43.14 | haooliveira | Hi Peoples! |
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| 19:17.00 | jonny87622 | I know this is the Asterisk room but I was hoping someone in here could help me configure a Polycom softkey to park a call. I have the soft key showing up on the phone it just is not doing what i need it to |
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| 19:35.42 | jonny87622 | any polycom experts in here? |
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| 19:48.15 | Synthase_ | People tend to charge for that type of black magic |
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| 19:51.00 | jonny87622 | @Synthase_ are you referring to polycom configs? |
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| 20:21.10 | Encrypt | Hi o/ |
| 20:21.22 | Encrypt | I'm trying to make chan_dongle work with my asterisk setup |
| 20:21.51 | Encrypt | However, loading chan_dongle fails because it doesn't find the appropriates ttyUSB |
| 20:22.03 | Encrypt | Has anyone played with chan_dongle, willing to help me? :) |
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| 20:32.04 | [TK]D-Fender | packs up to head home |
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| 21:04.55 | moe` | chan_dongle sounds almost pornographic. so that is to say, sorry, I can't help you. :) |
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| 21:29.57 | MaliutaLap | https://en.wikipedia.org/wiki/ZFS#Deduplication is pornographic |
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| 21:37.01 | moe` | so good. its quite simple but I just got internal->external and external->external extensions.conf working. already had external->internal. |
| 21:39.27 | moe` | so first, with all the landline numbers I have, it'll try all my SIP clients at the same time. failing that, it'll go to my cell phone. |
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| 22:00.08 | jms77 | I'm trying to setup mailbox/device sharing as described here:https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP |
| 22:00.27 | jms77 | After I've set up the configuraiton files, should I see instance1/instance2 when I do pjsip show endpoints from the CLI? |
| 22:00.40 | Encrypt | <moe`> chan_dongle sounds almost pornographic. so that is to say, sorry, I can't help you. :) // :D |
| 22:01.05 | Encrypt | moe`, It's a module which aim is to make a GSM gateway with asterisk |
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| 22:08.07 | moe` | Encrypt yeah I assumed it was something like that, still, it amused me. |
| 22:08.20 | Encrypt | :P |
| 22:08.34 | Encrypt | moe`, It seems to be a "switching" problem |
| 22:08.53 | Encrypt | Because at first, a USB modem is considered as a CD-ROM / Flash Drive |
| 22:09.06 | Encrypt | I'm playing with usb_modeswitch right now |
| 22:09.29 | moe` | yeah I've seen that here, get a WiMAX USB dongle, it shows as a CD or flash drive with the providers software (for 'doze) |
| 22:09.36 | moe` | install the software and some magic happens |
| 22:09.38 | moe` | and you get 'net |
| 22:19.25 | moe` | oddly, a GSM gate wouldn't help here. local number to local number can be 30-70 cents |
| 22:19.37 | moe` | skype to local number is about 11 cents |
| 22:20.11 | moe` | so in fact I typically send my local call out via the skype SIP connection |
| 22:20.25 | moe` | and just not use my local phone at all |
| 22:20.47 | moe` | (that's per minute) |
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| 22:23.07 | moe` | if calling a Canadian number, I send the call out via the flowroute SIP connection (they offer better rates) |
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