| 00:04.36 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 00:07.39 | *** join/#asterisk deeperror (~deeperror@d149-67-49-94.try.wideopenwest.com) |
| 00:09.47 | deeperror | In Asterisk 1.4 using Dial(). What exactly determines if we hear ringback or don't hear ringback when making SIP outbound dials? Namely UK telephone numbers. Some play the UK tone when we dial them until someone answers and other numbers seem to remain silent until someone answers or we get HANGUPCAUSE = NOANSWER |
| 00:21.40 | Kobaz | oh right |
| 00:21.48 | Kobaz | 1.8.2 seg faults on attended transfer |
| 00:21.53 | Kobaz | upgrades |
| 00:38.09 | *** join/#asterisk coppice (~chatzilla@9.160.232.220.dyn.pacific.net.hk) |
| 00:53.15 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 00:55.45 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 01:15.34 | tzanger | deeperror: the remote end would send that information back to your asterisk box |
| 01:16.47 | *** join/#asterisk manji (~manjiki@ppp-94-65-252-87.home.otenet.gr) |
| 01:18.32 | deeperror | tzanger: and i'm to understand it's not recommended to play ringing? So what's the recommended way to indicate a no answer to the caller? play recording? |
| 01:21.19 | tzanger | deeperror: no, the far side should generate a ringing sip message (not audio) which your asterisk should interpret and play the appropriate audio... I think |
| 01:21.26 | tzanger | it's been quite a while since I've worried about this |
| 01:21.41 | deeperror | like ringing-100 180? |
| 01:21.47 | tzanger | basically the far end gets to determine what to play, either by providing the audio itself or by providing appropriate sip messages |
| 01:22.01 | tzanger | yes, I believe so |
| 01:22.18 | deeperror | and the far end isn't my provider it's the callee co? |
| 01:22.40 | tzanger | in an ideal system, yes |
| 01:23.03 | tzanger | there are all kinds of screwily-configured providers :-) |
| 01:23.20 | deeperror | notice that :)) |
| 01:23.55 | *** join/#asterisk eugeneoden (~goden@63.133.138.10) |
| 01:24.05 | deeperror | i'm noticing it mostly on UK |
| 01:24.39 | deeperror | so i think i'll just have to use hangupcause and play a recording to our agents so they can mark the record correctly as invalid number is quite different from no answer |
| 01:25.35 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:28.04 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 01:30.16 | *** join/#asterisk coppice (~chatzilla@m121-203-193-0.smartone-vodafone.com) |
| 01:37.41 | tzanger | deeperror: you could try asking your ITSP why they're not doing the right thing, but make sure you have SIP logs that you can lead them through |
| 01:38.16 | deeperror | tzanger: ok thanks for the tips and time |
| 01:38.47 | tzanger | np, hope it helped |
| 01:46.22 | p3nguin | deeperror: Ringback ringing is provided by the far end device. If you are SIP/steve and you call SIP/mark, Mark's phone provides the ringing sound. |
| 01:48.34 | deeperror | p3nguin: will open ticket with my sip provider see what they say. Thanks! |
| 01:50.19 | p3nguin | I was having problem recently with no ring sound and I thought it was my provider. I added the r option on my dial commands to compensate. Turns out it was my channel driver not giving me rtp at the appropriate time. When I set my earlyrtp mode to progress rather than none, I got my ring sounds back... so I was able to remove the r option, finally. |
| 01:50.58 | JerJer | p3nguin: the r option is pretty evil |
| 01:51.39 | p3nguin | Yeah, but I needed to have ringing rather than silence while waiting on an answer. It was (assumed to be) necessary at that time. |
| 01:52.15 | p3nguin | As soon as I found out where the problem was, I got rid of it. I was able to sleep better that night. |
| 01:52.48 | *** join/#asterisk Smirker (~x@14-202-69-225.static.tpgi.com.au) |
| 01:52.54 | Smirker | hey. |
| 02:07.59 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 02:12.13 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 02:12.46 | tzanger | god I love stable internet. :-( |
| 02:16.48 | Smirker | i'm using IP auth with my voice server and i'm trying to set the callerid of outgoing calls. in sip.conf, fromuser/callerid are not defined. in my dialplan, i Set(CALLERPRES()=allowed) and then Set(CALLERID(num)=thephonenumber), finally I Dial(SIP/number@mytrunk). however the call always shows up as Private on the destination number. |
| 02:16.54 | Smirker | any ideas how to change this? |
| 02:18.20 | p3nguin | First, I would like to see you change your syntax to SIP/peer/extension. |
| 02:18.41 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 02:19.10 | p3nguin | If you want a caller ID to be sent, set a CALLERID(num) on the channel. Either with Set() or by the sip option callerid on the peer. |
| 02:20.24 | Smirker | why SIP/peer/extension? these are incoming calls, get pushed through an arbitrary set of prompts, then gets pushed out to an external number. |
| 02:20.43 | Smirker | i have set CALLERID(num) in the dialplan before Dial(), but it doesn't seem to be effective. |
| 02:22.34 | p3nguin | Why SIP/peer/extension? That's the standard way of sending a call to an extension via a known defined peer on the SIP channel technology. Using extension@peer is the standard way to handle a URI call to an unknown peer. And it can't be an inbound call if you're worried about setting the caller ID that goes -out-. |
| 02:23.44 | *** join/#asterisk bmg505 (~leon@196-209-120-116.dynamic.isadsl.co.za) |
| 02:24.15 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 02:24.34 | Smirker | i am forwarding to an undefined peer. it is an inbound call. customer calls voip number --> asterisk phone system --> gets forwarded to a store based on a DID. |
| 02:25.19 | Smirker | i want to set the callerid that goes out so that the store knows the callerid of the customer. however even setting the callerid as a number defined in asterisk isnt working. |
| 02:25.33 | Smirker | might call my provider. |
| 02:26.21 | *** join/#asterisk codefreeze-lap (~Steve_Mur@nv-69-68-103-77.sta.embarqhsd.net) |
| 02:29.21 | raden | paulc, hey |
| 02:30.43 | raden | Smirker, callerid by default passes through |
| 02:30.57 | raden | Smirker, I have the same setup and our callerids run right through |
| 02:31.32 | raden | someone calls in Presses 1 no pickup asterisk dials my cell I get the persons calling callerid |
| 02:31.38 | raden | Smirker, who is your provider |
| 02:32.09 | raden | Smirker, and there are very very few service providers that all you to set your own CID unless your a wholesaler |
| 02:34.28 | p3nguin | you're |
| 02:34.37 | p3nguin | unless *you're* a wholesaler |
| 02:36.41 | raden | p3nguin, thanks p3nguin, sorry bro just woke up im all F*ed up today :( |
| 02:37.01 | raden | I swear a mack truck ran me over in my sleep last night |
| 02:42.45 | p3nguin | This sucks a lot. There's apparently no software that opens visio vsd files. |
| 02:45.42 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 02:47.01 | Smirker | raden: provider doesn't let us support it. just emailed them. |
| 02:48.15 | raden | Smirker, thats normal |
| 02:48.16 | Smirker | kinda sounds like a bs excuse: "As far as I know and Sofian will confirm, Caller id isnât possible and I explained this to Jordan when we discussed the options, he said the number didnât matter. The main reason being is that is a number shows up they all report as the physical address registered as wickham st, this is illegal as if one of these numbers called emergency services they |
| 02:48.16 | Smirker | would attended the wickham st site. " |
| 02:48.58 | Smirker | we have our own sip trunk, our servers reside at their datacentre and we have several hundred incoming DIDs. thought they'd trust us enough to set remote party id. |
| 02:49.30 | raden | Smirker, callerid is becoming a bigger and bigger issue cause of so many people miss using it |
| 02:50.46 | p3nguin | Oh how I love being able to send any caller id I feel like sending. |
| 02:50.51 | Smirker | ah well. it doesn't bother me, just those above me. they'll get over it. :p |
| 02:51.00 | Smirker | p3nguin: that was the plan :p |
| 02:51.12 | Smirker | not maliciously, though. |
| 02:51.28 | p3nguin | Of course not. That would be illegal. |
| 02:53.22 | *** join/#asterisk zepmantra (~therock@112.201.37.40) |
| 02:54.52 | Freeaqingme | p3nguin, that's the nice thing of the year 2011. It's the year nobody did anything illegal |
| 02:55.21 | deeperror | p3nguin: do you have more info on the earlyrtp mode? |
| 02:55.58 | p3nguin | deeperror: I set it to progress and that was the end of it. What else did you want to know? |
| 02:56.15 | deeperror | where is this setting? |
| 02:56.28 | p3nguin | It's in my sccp.conf. |
| 02:56.29 | deeperror | what file? |
| 02:56.31 | deeperror | oh |
| 02:57.57 | deeperror | anything similar with sip? |
| 02:59.51 | p3nguin | I don't think so. I think SIP accepts early media always, so there's no setting for it. |
| 03:01.09 | Smirker | while we're all here |
| 03:02.32 | Smirker | i have an (almost) direct connection to our wholesalers SIP server through an incontended private network. i'm forcing the user of g711(u or a). is there anything to change in the default sip settings to reduce latency? |
| 03:04.43 | p3nguin | SIP is going to magically reduce latency? Wow, if you come up with something, I'm sure the masses will be happy to hear about this. |
| 03:05.38 | Freeaqingme | Smirker, I dont know how many hops that connection exists of, but I usually latency is dealt with at switches and connections between them |
| 03:05.50 | Smirker | ah okay. |
| 03:05.57 | Smirker | i was thinking more of reducing buffer sizes/etc |
| 03:07.38 | p3nguin | Do you have a jitter buffer turned on? |
| 03:07.45 | p3nguin | By default, it's off. |
| 03:07.46 | Smirker | turned off. |
| 03:08.20 | Smirker | i think i might just be being paranoid |
| 03:08.33 | p3nguin | What are your ping times? |
| 03:08.36 | Freeaqingme | what's your latency? |
| 03:08.52 | Smirker | well it's connected via a private network, but the servers are in different states :P |
| 03:08.53 | p3nguin | round trip, or one way, just specify. |
| 03:08.55 | deeperror | p3nguin: what about progressinband? |
| 03:08.58 | Smirker | 17ms consistantly |
| 03:09.06 | Smirker | round-trip. |
| 03:09.10 | Freeaqingme | the nice thing of computing is that usually everything can be measured. No need for paranoia (except for security) |
| 03:09.22 | deeperror | that only for receiving calls? |
| 03:09.40 | p3nguin | 17ms round trip is perfect for voice traffic. It doesn't start hurting until 100ms in one direction. |
| 03:09.57 | Smirker | my paranoia is i'm calling via mobile, that goes to sydney, call data goes back to brisbane (my server), sent back to sydney, send back to brisbane, and then out to a mobile again |
| 03:10.09 | Smirker | lag seems to be about 200ms-300ms |
| 03:10.31 | p3nguin | I'd say that delay comes from all the crap you have in the middle. |
| 03:10.43 | Smirker | yeah. |
| 03:14.59 | Severian | If I do a "iax2 show peers registered", does it show me the registrations I have made to other servers or the registrations other servers have made to this machine? |
| 03:18.57 | p3nguin | "iax2 show registry" shows what you are registered to. The one you're asking about should show only the peers that have a known address from the list that "iax2 show peers" produces. |
| 03:19.29 | p3nguin | So iax2 show peers shows all peers that you have configured, but not all of them have known addresses right now. |
| 03:19.41 | p3nguin | Your command reduces the output to known addresses only. |
| 03:20.01 | p3nguin | Not necessarily registered to you, either. |
| 03:20.28 | p3nguin | For example, you have your ITSP set with a static host rather than accepting regisrations. It will be in the list. |
| 03:20.36 | p3nguin | Hope that makes sense. |
| 03:22.39 | Severian | It makes some sense, but I don't understand the output I see. I have two asterisk boxes. They are each set to do an iax registry with the other. If I do iax2 show registry on the remote end, I see the local server with a column labeled Perceived showing <Unregistered>. |
| 03:23.01 | p3nguin | Compare it to iax2 show peers. |
| 03:23.03 | Severian | So, is it registered, since it shows there, or is it not? |
| 03:23.12 | p3nguin | I would say yes it is. |
| 03:23.34 | p3nguin | You could go to that host and issue iax2 show registry, to see what it is registering to. |
| 03:23.44 | p3nguin | It would show registered or something else. |
| 03:24.54 | Severian | The local end shows it as being registeres with Perceived showing the right IP address for the remote end. |
| 03:25.51 | p3nguin | If on the remove host the State says Registered, then the local side would show the IP address of the remote side. |
| 03:25.58 | p3nguin | s/remove/remote/ |
| 03:26.34 | p3nguin | Most people don't ever ask about that command. I'm surprised you're inquiring about it. |
| 03:27.36 | Severian | On the remote end, if I do iax2 show registry, I get "192.168.2.15:4569 N dallasoffi <Unregistered> 60 Request Sent" |
| 03:28.13 | p3nguin | So that remote has sent a request to register TO that host, but it never accepted. |
| 03:28.39 | p3nguin | show registry shows what you are registering *to*. |
| 03:29.05 | Severian | I want my calls to be routed from my local asterisk box through a vpn and then out from there to my ITSP. I had it all working yesterday and then today all I get on the local asterisk server is "WARNING[2384]: app_dial.c:1759 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)" |
| 03:29.13 | p3nguin | show peers registered shows any peer that you know an IP address for (could be static or registered to you). |
| 03:29.55 | p3nguin | Sounds like iax2 channel driver wasn't loaded. |
| 03:30.33 | Severian | I am trying to figure out where the failure is and I am a bit stumped. How would I check to see if the channel driver is loaded. |
| 03:31.14 | p3nguin | If iax2 commands work, that generally means it is loaded. You could additionally verify it with module show like iax. |
| 03:31.21 | Severian | show peers registered is not a valid command. I am running 1.6.2.16.2 |
| 03:31.44 | p3nguin | iax2 ... |
| 03:31.50 | p3nguin | I was abbreviating. |
| 03:32.15 | p3nguin | I thought you knew we were dealing in iax2, so I left it off. |
| 03:32.15 | Severian | ah, got it |
| 03:34.48 | Severian | On the local end, iax2 show peers registered shows no peers. module show liake iax shows chan_iax2.so with a use count of 0. |
| 03:35.51 | p3nguin | Does iax2 show peers also reflect that no peer has a known IP address? "... peers registered" should simply filter ONLY THOSE WITH KNOWN ADDRESSES. |
| 03:35.58 | Severian | Don't worry about it. I'll research some more here. I was mainly asking so I would have a clue as to which end was more likely to be the troublespot. |
| 03:36.51 | devdvd | ok, so I am having this problem with asterisk. it happens on 1.6.2.15 and 1.8.3.1 using different phones (a polycom soundpoint 321 and a device with sipdroid), happens on 2 different machines and on 2 different providers. The problem is this. When I dial externally to an auto attendant where I have to enter digits. It may take me 20+ tries before I can get the digits to go through correctly. This is on anything I dial that has an |
| 03:36.51 | devdvd | <PROTECTED> |
| 03:37.14 | Severian | iax2 show peers shows the peer with a Host showing as (Unspecified). That seems odd to me, since it should know the host. |
| 03:37.28 | p3nguin | Well, it doesn't. |
| 03:39.30 | Severian | That seems like it may be related to the problem then. Since calls need to go from the local machine to the remote one and the local end does not know where the remote end is. |
| 03:41.03 | p3nguin | Yep, that's a problem. |
| 03:47.17 | Kobaz | hey p3nguin |
| 03:47.30 | Kobaz | oh wait, you use sccp |
| 03:47.39 | Kobaz | i was gonna ask if you had a skinny.conf example for blf/sla |
| 03:47.51 | p3nguin | I sure don't. |
| 03:48.46 | *** join/#asterisk jetlag (~jetlag@pool-173-61-239-226.cmdnnj.east.verizon.net) |
| 03:52.48 | Kobaz | i how easy it is to set new configs on cisco phones |
| 03:52.58 | Kobaz | skinny reload and poof it's changed, no reboots |
| 03:54.12 | p3nguin | That's causing chan_skinny to reset the phones? |
| 03:55.18 | Kobaz | i guess |
| 03:55.19 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 03:55.28 | p3nguin | I don't think sccp reload causes the phones to reset. I think if I change something, I have to sccp reload and then sccp reset SEP000000000000 too. |
| 03:55.29 | Kobaz | er, s/i how/i love how/ |
| 03:55.40 | Kobaz | it reloads any phone that has a changed config |
| 03:58.48 | Kobaz | haha |
| 03:58.49 | Kobaz | uh oh |
| 03:59.02 | Kobaz | something is horribly wrong in 1.8.3.1 |
| 03:59.22 | Kobaz | sip to sip call is spawning two new threads every second |
| 03:59.35 | Kobaz | 184 threads listed. |
| 03:59.40 | Kobaz | 189 threads listed. |
| 04:02.07 | Kobaz | oh, that's weird |
| 04:02.22 | Kobaz | ael parsing might be a little weird/broken too |
| 04:04.55 | Kobaz | ah hah... so tilghman broke it :( |
| 04:17.25 | Kobaz | aww, transfers to skinny seg fault |
| 04:20.50 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:21.55 | *** join/#asterisk jetlag (~jetlag@pool-173-61-239-226.cmdnnj.east.verizon.net) |
| 04:22.06 | Corydon76-home | Kobaz: what did I break? |
| 04:23.41 | Kobaz | M18480 |
| 04:23.53 | Kobaz | that patch changed how macros are created from ael |
| 04:24.29 | Corydon76-home | How does that break AEL? |
| 04:24.31 | Kobaz | it's a major functional breakage for ael macros |
| 04:24.37 | Corydon76-home | No, it's not |
| 04:24.37 | Kobaz | here's a paste |
| 04:24.42 | Kobaz | sec |
| 04:25.20 | Corydon76-home | The only thing it can break is if somebody uses the *undocumented* entry point to how AEL implements subroutines |
| 04:25.47 | Corydon76-home | but if you use undocumented shit anywhere, it can break at any time |
| 04:26.09 | Kobaz | goto context, s |
| 04:26.19 | Corydon76-home | Don't do that |
| 04:26.25 | Kobaz | http://pastebin.com/s9uWc7UM |
| 04:27.04 | Kobaz | what's the approach to jump from extensions.conf then |
| 04:27.13 | Corydon76-home | We don't support that |
| 04:27.27 | Kobaz | say you have an ael macro defined, and you're pulling db config items as extensions.conf lines into dialplan |
| 04:27.40 | Corydon76-home | We recommend you use ONE or the OTHER, not a hybrid of both |
| 04:28.06 | Corydon76-home | Nevertheless, we created a new application in branch to deal with that |
| 04:28.11 | Kobaz | oh |
| 04:28.14 | Kobaz | well that's good |
| 04:28.22 | Kobaz | for now i just reverted that one patch |
| 04:28.22 | p3nguin | Wow, what makes sound files that I record on asterisk as .gsm, then convert to .ulaw, .wav, and .sln to playback like a bunch of garbage? |
| 04:28.47 | Kobaz | but i guess i should redo my stuff to not use any ael macros until the app is ready |
| 04:28.58 | Corydon76-home | p3nguin: gsm is compressed. Record as slin, then compress down to other formats |
| 04:29.07 | Corydon76-home | The app is already in branch |
| 04:29.14 | Kobaz | oh |
| 04:29.23 | Kobaz | which one? 1.8 |
| 04:29.28 | Corydon76-home | All of them |
| 04:29.38 | Kobaz | oh, hmm |
| 04:29.40 | Corydon76-home | 1.4, 1.6.2, 1.8, and trunk |
| 04:29.53 | Kobaz | ah okay |
| 04:29.56 | p3nguin | Do I Record() to .sln or .slin? file convert thinks it needs to be .sln, but Playback() thinks it's .slin |
| 04:30.01 | Kobaz | what's the name of it? :) |
| 04:30.21 | Kobaz | p3nguin: using file convert in asterisk? |
| 04:30.22 | Corydon76-home | I forget... AELsub, I think |
| 04:30.28 | p3nguin | kobaz: yes |
| 04:30.30 | Kobaz | that sounds awesome |
| 04:30.42 | Corydon76-home | p3nguin: any of them. |
| 04:31.14 | Kobaz | p3nguin: i've never had a problem converting gsm/ulaw/wav back and fourth |
| 04:31.20 | Corydon76-home | p3nguin: .sln or .raw |
| 04:31.23 | Kobaz | is the file playable with Playback before you converted it? |
| 04:31.39 | p3nguin | Wait, I was wrong. I'm somehow using Record(mymessage:wav) |
| 04:31.47 | p3nguin | I thought it was gsm. |
| 04:31.56 | Corydon76-home | p3nguin: then it's already recording in slin |
| 04:32.09 | Corydon76-home | .wav==16-bit audio |
| 04:32.11 | p3nguin | I've messed up something unknowingly. |
| 04:33.15 | p3nguin | I don't even know why I have the .gsm files. This doesn't make sense to me. |
| 04:33.19 | Corydon76-home | I'm sure if you ask someone nicely at Digium, you can take your 48kHz raw audio files and have them downconverted to every format that Asterisk supports |
| 04:33.59 | p3nguin | I better record another sound clip and see what it's doing, because I seem to end up with a .gsm file, but the dialplan says it is recording to wav. |
| 04:34.01 | Corydon76-home | p3nguin: because having audio in every format saves you CPU when it comes to playback time |
| 04:35.21 | p3nguin | I screwed up. I renamed a .wav file to .gsm thinking I had been recording .gsm files! |
| 04:35.36 | p3nguin | I didn't notice the extension of the original file name. |
| 04:36.03 | p3nguin | So now I get to re-record some voice prompts because I wasn't paying attention. |
| 04:36.18 | Kobaz | oh nice |
| 04:36.30 | Kobaz | i fixed a skinny transfer crash in 1.8.3.1 (i think) |
| 04:37.20 | Kobaz | Corydon76-home: yeah I didn't know that the 's' exten made by macros was an uncommented unstable thing to rely on |
| 04:37.41 | Kobaz | Corydon76-home: so i went from 1.8.2 to 1.8.3.1 and all of a suddon none of my jump points existed |
| 04:39.54 | p3nguin | So recording to .wav is the same as .sln? |
| 04:41.47 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 04:43.31 | Kobaz | Corydon76-home: quick question while you might still be here... how doable is it to run dialplan while also bridging audio to different peer |
| 04:43.46 | Kobaz | ie... something like the U option to Dial |
| 04:44.29 | Kobaz | i want to run some dialplan when the dialed party answer()s, but i know for certain that i want the audio to start (i'm not going to drop the call)... so i'm thinking i can do something like StartPeerMedia() or whatever |
| 04:45.03 | Kobaz | it will start passing audio but i can keep doing my diaplan processing which may take a while, otherwise the call takes a noticable amount of time to start passing audio |
| 04:45.14 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-vxcsdtrnjolkkqlz) |
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| 04:47.36 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 04:53.31 | Kobaz | mmm, another transfer to skinny crash |
| 04:53.38 | Kobaz | has to wait til morning |
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| 04:57.02 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 04:59.46 | *** join/#asterisk CodKet9 (~ketchuphe@24-247-90-148.dhcp.bycy.mi.charter.com) |
| 05:01.30 | *** join/#asterisk jetlag (~jetlag@pool-173-61-239-226.cmdnnj.east.verizon.net) |
| 05:02.33 | CodKet9 | I have asteriskNOW using the freePBX gui running on a virtual machine and I am trying to connect any SIP or IAX softphone to an extension on my asterisk server but i am not having any luck. |
| 05:03.19 | ChannelZ | #freepbx |
| 05:03.29 | CodKet9 | thank you |
| 05:04.09 | ChannelZ | you can look at the console and maybe get an idea of what is going on but if it's a config issue on the * side you have to deal with FreePBX |
| 05:07.15 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
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| 05:41.43 | *** part/#asterisk CodKet9 (~ketchuphe@24-247-90-148.dhcp.bycy.mi.charter.com) |
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| 06:54.30 | oquidave | hello people |
| 06:55.04 | oquidave | am configuringa digium TDM card for asterisk |
| 06:55.37 | oquidave | am connecting the 4 port card to a T1 line |
| 06:55.50 | oquidave | i have also already installed the dahdi drivers |
| 06:56.19 | oquidave | now when i type...dahdi show status, i get RED alarms across all the ports |
| 06:56.22 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:56.33 | oquidave | what does this mean? thanks anyone please |
| 06:57.47 | oquidave | The port connected to the T1 is showing RED lights like all the oother 3 that are not...what should i do? |
| 06:59.14 | oquidave | anyone please? |
| 07:12.27 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 07:19.17 | kaldemar | oquidave: check your cable. |
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| 07:28.56 | benngard | i am using Originate in the dialplan to send a fax, works fine, but i cant figure out howto set/change clid, the cdr for a call looks like "2011-03-17 08:09:10+01 | unavailable | unavailable | 0027127501", any knows howto change the clid? |
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| 07:32.34 | oquidave | @kaldemar....what do mean check my cable |
| 07:33.08 | oquidave | its plugged in..from the T1 to one port on the TDM card |
| 07:33.36 | oquidave | what exactly should i hceck with the cable |
| 07:34.25 | Aven | Help please, I use Skype for Asterisk but the unit in what there can be a problem doesn't boot normally? * 1.6.2.13, sfa 1.1.2-1.1.4 |
| 07:34.26 | Aven | http://pastebin.com/wLvsAX3E |
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| 07:45.51 | *** part/#asterisk clive- (~pirch@41.146.161.201) |
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| 07:49.45 | schmidts | good morning |
| 07:53.47 | wdoekes2 | morning |
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| 08:00.15 | kaldemar | benngard: use a local channel, then you can do pretty much what ever you want in the extension. |
| 08:02.44 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 08:03.22 | kaldemar | oquidave: that the cable is properly connected and the wiring is correct, start with that. |
| 08:06.21 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 08:06.39 | joobie | hey guys.. anyone got a guide on setting up the polycom ip 7000 for asterisk? |
| 08:15.45 | oquidave | @kaldemar, the cable(RJ45) is properly connected and wiring is fine as far as i know |
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| 08:18.08 | kaldemar | oquidave: 1,2 and 4,5? is the other end up and working? |
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| 08:29.44 | oquidave | @kaldemar what end? |
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| 08:36.36 | kaldemar | oquidave: the one that is not your asterisk box... |
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| 09:17.20 | tech__ | hello |
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| 10:40.59 | oquidave | \ |
| 10:41.59 | *** join/#asterisk eject_ck (5fd7edde@gateway/web/freenode/ip.95.215.237.222) |
| 10:42.03 | eject_ck | Hi all |
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| 11:18.16 | *** join/#asterisk gentoo_fun2 (seb@jet.bayhost.net) |
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| 11:19.18 | gentoo_fun2 | so i put deny=0.0.0.0/0.0.0.0 permit=192.168.1.0/255.255.255.0 alwaysauthreject=yes |
| 11:19.25 | gentoo_fun2 | on all my user/peer entries for my phones |
| 11:19.39 | gentoo_fun2 | but somehow script kittens still are able to attempt logins here |
| 11:20.12 | gentoo_fun2 | Registration from '"1951991620"<sip:1951991620@.. failed for '69.65.110.72' - No matching peer found |
| 11:20.17 | gentoo_fun2 | i get quite a few of these a day |
| 11:20.28 | gentoo_fun2 | been blocking with iptables, i will eventually use fail2ban |
| 11:20.47 | gentoo_fun2 | but i wanted to disallow access alltogether with asterisk? |
| 11:21.24 | fauxalliance | fail2ban is great |
| 11:26.59 | kaldemar | gentoo_fun2: deny/permit won't prevent incoming registration attempts, it only prevents the defined addresses to match the peer in question. |
| 11:27.11 | gentoo_fun2 | oic |
| 11:27.15 | fauxalliance | i.e. use them both |
| 11:27.21 | gentoo_fun2 | so they can continue to attempt |
| 11:27.28 | gentoo_fun2 | it just will always fail |
| 11:27.37 | gentoo_fun2 | even if they guess peer/user pass |
| 11:27.41 | kaldemar | gentoo_fun2: asterisk will still handle the requests, just block them in your firewall. |
| 11:27.51 | gentoo_fun2 | yea ive been doin that |
| 11:28.22 | gentoo_fun2 | is there anything else i should be doing? |
| 11:29.29 | fauxalliance | other than something like portsentry and / or fail2ban and your tightened policies... you should be fine |
| 11:29.58 | fauxalliance | else... monitor and block as you have been doing |
| 11:30.48 | gentoo_fun2 | aye im learning as i go |
| 11:30.59 | gentoo_fun2 | i cant imagine why they even want access, to make long distance calls? |
| 11:31.24 | fauxalliance | grey routes, free calls, harass other people... who knows |
| 11:31.34 | kaldemar | gentoo_fun2: http://svn.digium.com/svn/asterisk/tags/1.8.0/README-SERIOUSLY.bestpractices.txt |
| 11:32.16 | gentoo_fun2 | hmm okay |
| 11:32.21 | gentoo_fun2 | i can see ive already made mistakes |
| 11:32.25 | gentoo_fun2 | ill read it thx |
| 11:33.39 | tzanger | kaldemar: that's awesome |
| 11:35.32 | fauxalliance | likes the MAC address idea.... then runs off to implement it... |
| 11:37.49 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 11:41.52 | eject_ck | does it make sense to move inbound call to voicemail if calee not respond for 60 seconds ? |
| 11:42.11 | tzanger | eject_ck: isn't that generally how voicemail should work? |
| 11:43.05 | eject_ck | tzanger: yes I think |
| 11:43.22 | tzanger | I think so too |
| 11:43.30 | tzanger | although 60s is an awfully long time |
| 11:45.44 | gentoo_fun2 | yes |
| 11:45.48 | gentoo_fun2 | ours is like 23s |
| 11:45.53 | gentoo_fun2 | idk how we got to that point |
| 11:46.03 | gentoo_fun2 | something about my boss saying "try and make it 4 rings" lolo |
| 11:46.15 | *** join/#asterisk salimb (~3laz3r@83-244-177-2.cust-83.exponential-e.net) |
| 11:49.26 | tzanger | gentoo_fun2: well a normal (north american) ring cadence is 2s ring, 4s silence. 24/6 = 4 rings |
| 11:50.02 | gentoo_fun2 | o wow |
| 11:50.10 | gentoo_fun2 | i never bothered to think of that |
| 11:51.25 | gentoo_fun2 | how do you do a "graceful" asterisk restart? |
| 11:57.02 | *** join/#asterisk defswork (~andy@mx1.3gcomms.co.uk) |
| 11:57.21 | fauxalliance | <PROTECTED> |
| 11:58.19 | fauxalliance | waits for no calls... |
| 11:58.34 | fauxalliance | 'restart gracefully' STOPS taking calls, and restarts when all are finished |
| 12:01.01 | *** join/#asterisk ripon (~ripon@78-86-161-207.zone2.bethere.co.uk) |
| 12:01.48 | ripon | a really easy problem i could do with an answer - save me looking on google.... |
| 12:02.58 | ripon | when i dial exten 800 ,,, i want to ring a pre defined telephone number using my sip voipprovider- which is already registered opn my asterisk box |
| 12:03.59 | fauxalliance | ripon, exten => 800,1,Dial(SIP/1234@itsp.ws) ? |
| 12:04.22 | ripon | thanks fauxalliance- |
| 12:04.50 | ripon | i only ever use my setup to recieve calls - never make them ... |
| 12:04.59 | ripon | till now ,,, cheers |
| 12:05.43 | fauxalliance | ;) |
| 12:05.48 | ripon | if my provider is sip.voipgain.com |
| 12:06.01 | fauxalliance | cares not to care |
| 12:06.07 | fauxalliance | sorry |
| 12:06.09 | ripon | then it would be sip/telephone number@sip.voipgain.com? |
| 12:06.28 | fauxalliance | ripon, thats how URI based dialing goes |
| 12:06.53 | ripon | once again. thx |
| 12:08.56 | eject_ck | it works in another way for me |
| 12:11.29 | eject_ck | tzanger: my extension is |
| 12:11.30 | eject_ck | exten => _3XX,1,Dial(SIP/${EXTEN},30,tTw) exten => _3XX,n,VoiceMail(${EXTEN}@default) exten => _3XX,n,PlayBack(vm-goodbye) exten => _3XX,n,Hangup() |
| 12:16.10 | *** join/#asterisk skrusty (~ben@83.166.169.221) |
| 12:16.15 | skrusty | afternoon all |
| 12:16.53 | fauxalliance | eject_ck, indeed! |
| 12:17.36 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
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| 12:32.27 | eject_ck | what's wrong with it ? |
| 12:32.57 | eject_ck | when I'm not pick call for 30 seconds I just head PlayBack(vm-goodbye) and this it |
| 12:33.01 | eject_ck | no voicemail prompt |
| 12:33.30 | Dovid | hi. when asterisk gets a sip timer message say for 1800 asteirsk sends a re-invite after 9000. anyone know why this is ? It seems asterisk is sending a re-invite for half the time. |
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| 12:34.37 | gentoo_fun2 | /etc/init.d/asterisk reload |
| 12:34.38 | gentoo_fun2 | * Forcing asterisk to reload configuration... |
| 12:34.45 | gentoo_fun2 | same as gracefuk? |
| 12:34.48 | gentoo_fun2 | l |
| 12:35.41 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:36.19 | SiNGLer | check the script, maybe it is same as "reload" in cli |
| 12:36.57 | *** join/#asterisk jnicola (~jnicola@host2.186-124-123.telecom.net.ar) |
| 12:39.32 | devdvd | gentoo_fun2, yes, /etc/init.d/asterisk reload is the same as doing module reload from the cli (at least according to the init script) |
| 12:41.40 | gentoo_fun2 | module reload also reloads confs? i am simply doing sip.conf etc edits |
| 12:41.50 | gentoo_fun2 | seems like restarting asterisk is very dumb |
| 12:41.54 | gentoo_fun2 | for something so simple |
| 12:42.14 | devdvd | yes |
| 12:42.32 | *** join/#asterisk bullium (~wbradshaw@216.68.250.30) |
| 12:42.37 | devdvd | you can also do asterisk -rx "module reload" from the prompt |
| 12:42.50 | devdvd | not from the cli but from your os shell |
| 12:42.59 | gentoo_fun2 | anyway to do in cli? |
| 12:43.05 | devdvd | module reload |
| 12:43.07 | gentoo_fun2 | thx |
| 12:43.15 | gentoo_fun2 | ill do that from now on |
| 12:43.17 | gentoo_fun2 | thx alot |
| 12:43.19 | devdvd | ok |
| 12:43.21 | devdvd | np |
| 12:44.54 | kaldemar | gentoo_fun2: if you only modify sip.conf, use "sip reload" command. no need to reload all modules if you only change settings for one. |
| 12:45.34 | gentoo_fun2 | can u do that with any conf? extensions reload? |
| 12:46.58 | devdvd | dialplan reload |
| 12:49.58 | kaldemar | gentoo_fun2: many modules register CLI commands that can be used. those that don't can also be reloaded with "module reload". see "core show help module reload". |
| 12:50.21 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 12:51.43 | gentoo_fun2 | ahh thx |
| 12:57.44 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 12:58.22 | adyn | Hello, is it possible without patching to copy the selections make in one menuselect to another compile of asterisk? and if so, what file(s) need to be copied? |
| 12:59.11 | adyn | same branch, 1.6.2.0 -> 1.6.2.17.1 |
| 13:00.21 | *** join/#asterisk serafie (~erin@nat/digium/x-gneilqjhxzawpibf) |
| 13:05.41 | devdvd | adyn, im not 100% sure but you could try menuselect.makeopts and menuselect.makedeps |
| 13:06.41 | devdvd | not sure if thats the full list or just the ones selected though. |
| 13:08.53 | *** join/#asterisk paulebeinlich (~paulebein@port-87-234-219-124.static.qsc.de) |
| 13:08.58 | paulebeinlich | hi |
| 13:10.18 | *** join/#asterisk coppice (~chatzilla@9.160.232.220.dyn.pacific.net.hk) |
| 13:10.40 | paulebeinlich | i am a newbie looking for help with asterisk 1.6+sangoma+dahdi....could someone give me a hint to solve my problem with a d-channel that will not work |
| 13:11.22 | adyn | devdvd: those looked the most promising, I'll give it a try. Thanks. |
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| 13:21.47 | rcaskey | So am I understanding that it is possible now to get traffic digitally from your upstream phone provider so that you don't need any special equipment at all in your server closet? |
| 13:21.58 | *** join/#asterisk goden2 (~goden@70.158.103.10) |
| 13:27.36 | tzafrir | paulebeinlich, hmm... just ask |
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| 13:32.07 | Dovid | hi. when asterisk gets a sip timer message say for 1800 asteirsk sends a re-invite after 9000. anyone know why this is ? It seems asterisk is sending a re-invite for half the time. |
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| 13:41.30 | ruben23 | hi guys i have a voip carrier grnvoip- that i need to ssetup my every call to used a route they designate --> http://pastebin.com/hTTc0ikh ---> but the problem is it wont dial at all and it wont recieve incoming calls, this is my setup --> http://pastebin.com/XdzrNL5N |
| 13:43.02 | ruben23 | any idea guys..? |
| 13:43.30 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 13:43.53 | Dovid | looking |
| 13:44.34 | ruben23 | the issue also for this is when i dial 1 + 10 digit number for us it would say invalid extensions becasue of the prefix- i should be able to remove the prefix somehow |
| 13:44.38 | Dovid | ruben23: Is the issue making ot getting calls ? |
| 13:45.00 | Dovid | ruben23: When you make a call out you dial 1+10 digit number ? |
| 13:45.12 | ruben23 | Dovid: getting calls and incoming calls also- also manual dial |
| 13:45.20 | ruben23 | yes |
| 13:45.43 | *** join/#asterisk goden2 (~goden@70.158.103.10) |
| 13:45.47 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 13:46.04 | Dovid | ruben23: for outbound try this: http://pastebin.com/tMB06Gv6 |
| 13:46.09 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:46.42 | Dovid | the dial plan is what aserisk gets. when you a call you are calling a 10 digit number. you need to have asterisk add the prefix if u do not want to dial it every time |
| 13:47.34 | ruben23 | Dovid: thank you so much, how about my incoming calls |
| 13:47.58 | paulebeinlich | tzafrir i try to explain a little bit |
| 13:48.32 | Dovid | ruben23: Please post sip.conf and extensions in full (with out passwords please) |
| 13:48.37 | paulebeinlich | i installed wanpipe and dahdi everythind looks nice |
| 13:49.02 | paulebeinlich | wanpipe1 | AFT TE1 | N/A | Connected | |
| 13:49.15 | ruben23 | Dovid: ok |
| 13:49.15 | paulebeinlich | wanpipe1 | N/A | A101/1D/A102/2D/4/4D/8| 16 | 4 | 1 | N/A | 0 | |
| 13:50.24 | paulebeinlich | wanpipemon -i wXg1 -c Ta -> Rx Level : > -2.5db |
| 13:50.42 | Dovid | paulebeinlich: Please do not flood. please use Paste Bin |
| 13:50.42 | paulebeinlich | no e1 alarm |
| 13:50.42 | Dovid | ~pb |
| 13:50.43 | 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. |
| 13:50.59 | paulebeinlich | sry |
| 13:54.25 | *** join/#asterisk asteriskator (97c8ea7b@gateway/web/freenode/ip.151.200.234.123) |
| 13:54.32 | asteriskator | Hi all |
| 13:54.35 | rcaskey | I spent a few hours last night skimming "The Book" and it cleared up some things for me but if you are willing to throw away all the non-SIP stuff in your building and shopping for a new telephony provider upstream, what can you do in terms of minimizing actual equipment in your closet? |
| 13:55.00 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:55.33 | *** join/#asterisk drift-_ (18914e7d@gateway/web/freenode/ip.24.145.78.125) |
| 13:55.48 | asteriskator | Does any one has some knowledge for setting Asterisk to authenticate using Radius?? |
| 13:56.08 | paulebeinlich | [Mar 17 14:55:36] WARNING[14822]: chan_dahdi.c:4160 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! --> http://pastebin.com/ze7d5hwk |
| 13:58.11 | devdvd | rcaskey, if you are talking about the end goal then the only thing you will technically need is your asterisk server(s) and any equipment used to connect up to the network (switches/routers/etc) |
| 13:58.48 | asteriskator | Any one RADIUS Client on Asterisk? I think I need some guru on this matter |
| 13:59.06 | devdvd | using a complete sip stack is 100% ip based so you won't need any extra equipment to connect the phones |
| 14:00.07 | rcaskey | devdvd, yes |
| 14:00.40 | rcaskey | devdvd, but I'd like to keep my old phone numbers and ideally have whoever maintains the line into the building also turn the sip traffic into POTS |
| 14:00.57 | rcaskey | and of course it should be cheaper, more reliable, make my life easier, and give me a free pony |
| 14:03.23 | *** join/#asterisk volga629 (~slava@host7.pythian.com) |
| 14:03.44 | volga629 | [Mar 17 09:37:28] VERBOSE[5190] logger.c: -- Channel 0/21, span 1 got hangup request, cause 16 |
| 14:03.55 | volga629 | what mean cause 16 ? |
| 14:04.11 | volga629 | thank you in advance |
| 14:04.40 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:05.13 | devdvd | well, as far as keeping your old phone numbers, you can port those, as far as maintaining the lines coming in. If you want to use your existing lines then you can use fxs ports to connect them to asterisk (that would allow you to keep your current configuration on the outside and move to a sip based model on the inside) |
| 14:05.42 | asteriskator | The biggest secret in Asterisk... Setting a RADIUS Client for authentication. Any one with some experience?? |
| 14:06.53 | devdvd | asterisk is very flexible in what it can use (SIP, IAX, PSTN, etc) |
| 14:08.03 | devdvd | so you can have all sip on the inside and pstn on the out or you can have sip on the outside and all pots in or a mixture of the things (sip/pstn on the out and in (and even iax2, sccp, h.323, etc) |
| 14:08.43 | rcaskey | devdvd, I'd like to get rid of all the specialized hardware if feasible, I'm fairly clear on basically how a more conventional setup would go |
| 14:08.45 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:08.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:09.35 | rcaskey | and I'v got the basic idea of what would need to happen on the inisde |
| 14:09.43 | devdvd | well, to go full sip on the inside and out you would need to have sip based phones (or ata's that will convert your current phones to sip |
| 14:10.09 | devdvd | but as far as the outside. hook your asterisk box to the internet and get you an itsp |
| 14:10.17 | devdvd | ~itsp |
| 14:10.17 | 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. |
| 14:10.35 | rcaskey | ~itsplist-us |
| 14:10.35 | 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 |
| 14:11.23 | devdvd | i also use one called Velocity which is more of a business setup, not bad prices and their support is excellent. |
| 14:11.40 | rcaskey | devdvd, we aren't particularly price sensitive |
| 14:11.54 | rcaskey | we don't love to pay more than we have too, but it is very important to have good authoratative service |
| 14:12.24 | devdvd | yea, finding a good itsp can be pretty trying |
| 14:13.16 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:13.52 | rcaskey | devdvd, are those same services likely to be available from our current phone provider? |
| 14:14.11 | devdvd | maybe but usually not |
| 14:14.45 | devdvd | most pstn providers aren't in the business of providing sip trunks, cuts into their profit margin |
| 14:14.50 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:15.26 | rcaskey | http://smallbiz.att.com/OSB/Idea-Exchange/Idea-Exchange-Detail.page?type=LiveSite:News&dcr=templatedata/LiveSite/News/data/SIP_Trunking_Enabling_Enhanced_Communications.xml&contentId=fwplfjo2 |
| 14:15.40 | *** join/#asterisk codefreeze-lap (~Steve_Mur@wsip-24-234-181-18.lv.lv.cox.net) |
| 14:16.40 | pabelanger | volga629: normal clearing |
| 14:18.51 | devdvd | well, if you have att as your provider and they provide sip trunks then great, you can stay with them for that service, however, it won't come in over the current phone lines you have into the building |
| 14:19.09 | rcaskey | devdvd, we got t1s and i'm looking to move to fiber at some point |
| 14:19.13 | devdvd | unless you have a t1 |
| 14:19.42 | rcaskey | AT&T gave me a call w/ fiber and I said "is it new? great. call me back in 2 years" |
| 14:19.55 | rcaskey | So they and some other companies are all hammering out the kinks :P |
| 14:20.20 | devdvd | but even at that...t1 will only do 1.54Mb, depending on how many calls you are doing at once that probably wont be near enough. |
| 14:20.37 | devdvd | especially if you are using that t1 for other data traffic |
| 14:20.51 | rcaskey | devdvd, I believe right now we have 23 voice channels |
| 14:21.05 | rcaskey | 1 for voice 1 for data |
| 14:21.50 | rcaskey | so as a transitional step we could move to sip, later bond them and do QOS |
| 14:21.50 | volga629 | pabelanger: so them mean some body press * |
| 14:21.57 | rcaskey | then jump ship to fiber if they offer us a better deal |
| 14:22.05 | volga629 | for exit |
| 14:22.42 | volga629 | pabelanger: where I can see some documentation about it |
| 14:23.04 | *** join/#asterisk roxdragon (~roxdragon@unaffiliated/roxdragon) |
| 14:23.09 | roxdragon | hi all |
| 14:23.11 | roxdragon | hjelp |
| 14:24.27 | pabelanger | volga629: no, this code is generated by the network. What interface is this? ISDN? |
| 14:24.34 | roxdragon | I have a nokia n95 and a Linksys SPA3102. When you connect the Nokia to the PBX (outside) is recorded but does not make me call the phone connected to the ATA |
| 14:25.17 | volga629 | pabelanger: yes on server we have PRI |
| 14:25.57 | volga629 | user start complain about dropping conference calls |
| 14:26.16 | roxdragon | help :( |
| 14:26.28 | pabelanger | volga629: you need to look into ISDN cause codes |
| 14:26.33 | pabelanger | ~ask |
| 14:26.33 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:26.38 | pabelanger | roxdragon: ^ |
| 14:27.21 | no1peanut | Hi - I want a caller to be able to trigger audio playback via dtmf into a conversation between 2 ppl. Would I need to make a conference or could I do it another way ? |
| 14:27.34 | roxdragon | pabelanger, show dialplan? |
| 14:27.45 | pabelanger | ~collectdebug |
| 14:27.46 | infobot | rumour has it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 14:27.53 | pabelanger | roxdragon: ^ |
| 14:28.43 | pabelanger | no1peanut: DYNAMIC_FEATURES in features.conf |
| 14:29.06 | volga629 | <pabelanger: thank you I found those code |
| 14:30.00 | no1peanut | pabelanger, thx .. will look at it :) |
| 14:34.00 | roxdragon | pabelanger, http://paste.ubuntu.com/581605/ |
| 14:35.15 | pabelanger | roxdragon: read the first line in the debug log |
| 14:35.21 | pabelanger | that is your issue |
| 14:35.38 | pabelanger | Call from '402' to extension 'gianni91.homeunix.com' rejected because extension not found. |
| 14:35.42 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 14:37.00 | roxdragon | yes |
| 14:37.05 | roxdragon | how solve? |
| 14:37.24 | *** join/#asterisk jkprg (~jarda@ip-62-245-93-150.net.upcbroadband.cz) |
| 14:37.45 | jkprg | Hi. I have multicast audio stream in my network. How can I configure asterisk to allow people to call specific extension to listen that stream? Thx |
| 14:37.52 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 14:38.12 | pabelanger | roxdragon: Have you defined 'gianni91.homeunix.com' as a valid sip peer (sip.conf)? |
| 14:39.21 | *** join/#asterisk sourcode (~code@ppp-58-8-86-110.revip2.asianet.co.th) |
| 14:40.06 | roxdragon | no.. the peer is 402 (NOKIA N95) When try a connect from extern to PBX ... |
| 14:40.17 | roxdragon | http://paste.ubuntu.com/581606/ this is extensions.conf |
| 14:41.29 | roxdragon | pabelanger, http://paste.ubuntu.com/581607/ This is SIP.CONF |
| 14:42.18 | roxdragon | gianni91.homeunix is PBX server domain |
| 14:42.41 | pabelanger | then looks like a SIP registration issue |
| 14:43.30 | pabelanger | also: exten => _X.,1,Dial(SIP/pstn/${EXTEN}) |
| 14:43.53 | pabelanger | same change for daEutelia |
| 14:45.07 | roxdragon | eutelia it's ok |
| 14:45.43 | roxdragon | but not work NOKIA(from external) > PBX > SPA3102(telephone) |
| 14:53.54 | asteriskator | Good Morning, Does any one can give me some hints for setting Radius with Asterisk? Thanks. |
| 14:53.55 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 14:53.55 | kaldemar | roxdragon: you didn't paste the interesting part of the sip debug, i.e. the INVITE message. |
| 15:08.35 | roxdragon | kaldemar, nokia (external sip) > PBX > SPA3102 don't work... but SPA3102(Telephone) > PBX >NOKIA (external) Work |
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| 15:20.26 | kaldemar | roxdragon: yes, that's clear. it's the details that are missing. looks like you somehow manage to dial "gianni91.homeunix.com" instead of a number on the N95. |
| 15:20.35 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:20.57 | asteriskator | I am willing to pay some support for some RADIUS Client configuration instructions for authenticating with Asterisk and radiusclient-ng. Some one? Thanks |
| 15:23.46 | roxdragon | I can show you what to fix? |
| 15:25.20 | Dovid | hi. when asterisk gets a sip timer message say for 1800 asteirsk sends a re-invite after 9000. anyone know why this is ? It seems asterisk is sending a re-invite for half the time. |
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| 15:31.27 | kaldemar | roxdragon: dial a number that you have in your dialplan. |
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| 15:42.28 | *** join/#asterisk Zhad (~tom@host-1.art-it-services.co.uk) |
| 15:44.35 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 15:44.42 | Zhad | We have a tannoy-style system here, when someone dials the PA extension, it Page()s phones that can be paged and a speaker system that's driven from chan_console |
| 15:45.20 | Zhad | Can anyone think of a way that I can get the message to repeat over chan_console when the caller hangs up? |
| 15:46.01 | Zhad | I suppose I could get it to monitor the call to a specified file, then on hangup get a system call to play it. |
| 15:46.08 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:46.37 | Zhad | But I'd only want that to trigger if that extension is hung up, not everything else in the context |
| 15:50.32 | asteriskator | Does anyone see my messages in the room? |
| 15:50.33 | kaldemar | Zhad: originate a call in the hangup extension (h). |
| 15:50.42 | kaldemar | asteriskator: sure. |
| 15:50.53 | asteriskator | oohh, ok. Thanks |
| 15:53.32 | asteriskator | Does any one know about Asterisk and Radius? Thanks |
| 15:55.41 | Zhad | kaldemar> I could set MONITOR_EXEC to a perl file that plays back |
| 15:55.54 | Zhad | the recording produced my Monitor |
| 15:56.09 | Zhad | or even just set it to something that can play it |
| 15:58.04 | jaytee | asteriskator, try this link for documentation: http://www.asterisk.org/docs there is info there for Radius. Google is also your friend |
| 16:00.56 | asteriskator | Jaytee, I put the Asterisk to work accounting CDRs using Radius, but the documentation for Authenticating SIP using radius is the biggest secret around, no one documented it. |
| 16:01.44 | kaldemar | Zhad: or just exten => h,1,Originate(Console/dsp,app,Playback,path/to/your/file) |
| 16:04.02 | asteriskator | The radius auth subject looks to be no so popular. |
| 16:04.36 | asteriskator | Has someone connected an Asterisk to a VoiceMaster billing platform for AAA? |
| 16:05.19 | Zhad | But then every time any call is hung up it woudl play through console/dsp |
| 16:05.53 | *** join/#asterisk wiit (~wiit@188.19.8.148) |
| 16:06.05 | wiit | hello all |
| 16:06.18 | *** join/#asterisk luckman212 (~irc@pool-173-77-253-141.nycmny.fios.verizon.net) |
| 16:06.21 | jaytee | asteriskator, this link might also be helpful: http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html#AEN53 |
| 16:06.58 | wiit | can anyone tell me what is the maximal value of dialplan variable? |
| 16:07.08 | leifmadsen | you mean maximum length? |
| 16:07.14 | wiit | yes |
| 16:07.17 | leifmadsen | in 1.8 I think it is something like 2048 chars |
| 16:07.54 | wiit | humm...i have version 1.6 installed, does it matters? |
| 16:07.58 | kaldemar | Zhad: only if your dialplan structure makes it so. there's also options F and g for the Dial application in current asterisk versions that allow dialplan execution to continue in the next priority despite a hangup. |
| 16:13.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:13.55 | asteriskator | jaytee, thanks for the link. It looks like the asterisk needs to be patched in order to allow radius authentication. I hope that patch from 2007 can work with the Asterisk 1.6 I am using. |
| 16:16.14 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
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| 16:18.36 | jaytee | asteriskator, I doubt those patches are still relevant. As a reference that post might contain useful information but I'd post there asking about if new patches are available or have been included in 1.6 already. |
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| 16:34.43 | *** join/#asterisk cryptnix (~andrew@pool-98-115-168-26.chi01.btas.verizon.net) |
| 16:35.04 | cryptnix | hmm -- when setting up a dial plan ... to allow anything to be passed whats the best ? X|. ? |
| 16:39.34 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 16:42.53 | asteriskator | jaytee, good advise. As I know the CDR part was included and documented, and the authentication looks to be included also, but not documented. Probably I will need to read the source for the chan_sip.c for some hints. |
| 16:43.00 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 16:49.02 | leifmadsen | cryptnix: well, best is a relative term, but to match "anything" is _. (match 1 or more characters) however it should be qualified you should not use that without FILTER() |
| 16:49.55 | leifmadsen | please read the README-SERIOUSLY.bestpractices.txt file in your Asterisk source |
| 16:50.02 | cryptnix | ok |
| 16:50.44 | *** join/#asterisk moos3 (~moos3@cpe-74-75-157-202.maine.res.rr.com) |
| 16:56.33 | *** join/#asterisk weller3 (~weller@dsl-dynamic-209-43-10-246.iquest.net) |
| 16:59.21 | weller3 | does res_jabber have any type of timeout or keepalive controls? |
| 16:59.46 | *** join/#asterisk codefreeze-lap (~Steve_Mur@wsip-24-234-181-18.lv.lv.cox.net) |
| 16:59.54 | benngard | weired, i wrote a small dialplan app that receives a fax, convert it to pdf and mail it to me, works when i send the fax from a fax attached to asterik through an ata, works when i use a call file to send the fax, but i doesnt work over ooh323 :( |
| 16:59.59 | weller3 | esp while using tls behind nat? |
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| 17:25.35 | Kobaz | pokes Qwell |
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| 17:37.12 | ssureshot | I'm setting up a new system going from version 1.2 to 1.8... I cant seem to get my extensions to work.. I'm getting the following in my asterisk cli when I try to call... http://pastebin.com/NqmtthVV |
| 17:37.55 | ssureshot | any help is greatly appriciated |
| 17:38.25 | leifmadsen | ssureshot: not enough info -- looks like your SIP device isn't reachable or isn't responding -- look at the sip trace and make sure the device is registered |
| 17:39.46 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 17:40.03 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
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| 17:41.28 | ssureshot | leifmadsen: you got it... I need to figure out how to do a sip trace lol.. I assumed that since I was communicating with the cli and could see the phones try to dial that I was registered |
| 17:41.48 | leifmadsen | sip set debug on |
| 17:42.01 | leifmadsen | sip set debug ip 192.168.0.1 |
| 17:42.06 | leifmadsen | sip set debug off |
| 17:42.19 | *** join/#asterisk ppc (~ppc@cloudy.cmang.org) |
| 17:42.20 | leifmadsen | (first line is for all sip messages, 2nd one is for a specific IP) |
| 17:47.14 | moos3 | leifmadsen you havea minute ? |
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| 17:57.39 | ssureshot | hmm, I have no idea what I'm looking at on the sip debug, it looks good to me.. |
| 17:58.31 | *** join/#asterisk _zoom_ (~Administr@196.1.251.36) |
| 17:59.01 | _zoom_ | hello, can I playtones with run answer before it? |
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| 18:01.55 | _zoom_ | hello, I need to playtones with run answer? |
| 18:02.08 | *** join/#asterisk iulius (~iulius@adsl-217-3-19.asm.bellsouth.net) |
| 18:04.20 | bdonegan | Hi, I'm new to Asterisk and I've read through the Getting Started section of the documentation on asterisk.org. I'm trying to setup two soft phones that simply call one another by extensions. I'm using the Blink sip client on MacOSX and Windows. Both phones seem to register fine, but when a call is made, the other phone can answer the call, the bridge looks successful, and then the call is disconnected. I'm using version 1.8.3.2, and I've copied my |
| 18:04.31 | bdonegan | thank you for any help |
| 18:05.47 | emora | bdonegan: Is the asterisk machine and the two Blink endpoints on the same LAN? |
| 18:06.04 | bdonegan | yes they are |
| 18:07.42 | emora | Blink will allow you to call from one softphone to the other without going through Asterisk. Have you tried to make sure they're working correctly? |
| 18:08.06 | bdonegan | I have not, let me give that a try. |
| 18:08.40 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 18:09.07 | *** join/#asterisk drift_ (42e933ee@gateway/web/freenode/ip.66.233.51.238) |
| 18:09.10 | emora | Select Bonjour on the list of accounts then dial the IP address of the other machine |
| 18:11.00 | bdonegan | That works just fine. |
| 18:11.14 | bdonegan | Calling both ways. |
| 18:11.16 | emora | Are the Blink clients registering with Asterisk? |
| 18:11.50 | bdonegan | They both show up with correct ip addresses with: sip show peers |
| 18:13.03 | emora | When you try to dial the other extension what output do you see on the Asterisk console? |
| 18:14.01 | bdonegan | with verbose set at 3: == Using SIP RTP CoS mark 5 |
| 18:14.01 | bdonegan | <PROTECTED> |
| 18:14.04 | bdonegan | <PROTECTED> |
| 18:14.07 | bdonegan | <PROTECTED> |
| 18:14.09 | bdonegan | <PROTECTED> |
| 18:14.12 | bdonegan | <PROTECTED> |
| 18:14.15 | bdonegan | <PROTECTED> |
| 18:14.17 | bdonegan | <PROTECTED> |
| 18:19.09 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 18:21.17 | SuPrSluG | Trying to use the D option when dialing to connect to an IVR then go to voicemail. Seems the 8 in the mailbox number is not being recognized more often than not, causing it to fail to connect to voicemail box. Any way to adjust the SendDTMF tx gain or length ? |
| 18:23.18 | *** join/#asterisk roxdragon (~roxdragon@unaffiliated/roxdragon) |
| 18:24.44 | *** join/#asterisk ripon (~ripon@78-86-161-207.zone2.bethere.co.uk) |
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| 18:25.59 | Naikrovek | bdonegan: don't paste in here please |
| 18:26.34 | ripon | anyone use one of the betavoip providers succesfully ? im trying out voipgain.com ,,, bit tricky to make a call oyt,,, either with nokia sip client , or on asterisk- keep getting "congestion" ...using the providers own softfone- no problem ... any advice? |
| 18:27.02 | bdonegan | my apologies |
| 18:32.23 | yoda1410 | a |
| 18:38.17 | bdonegan | emora: my extensions.conf just has: exten => 6001,1,Dial(SIP/stewie,20) each way and Hangup() |
| 18:42.38 | *** join/#asterisk distributed (~root@66.241.104.83) |
| 18:43.07 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-qpcncvriirxdkbne) |
| 18:44.23 | Kobaz | oh, 1.8.3.2 is out already |
| 18:45.02 | bdonegan | sure is, http://www.asterisk.org/downloads |
| 18:45.22 | *** join/#asterisk kaii (~kh@ciphron.de) |
| 18:45.26 | Kobaz | * AST-2011-003: Resource exhaustion in Asterisk Manager Interface |
| 18:45.38 | Kobaz | is that the thing where it keeps spawning new threads? |
| 18:47.53 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 18:48.05 | *** join/#asterisk eduardonunesp (~eduardo@187.65.203.147) |
| 18:48.21 | eduardonunesp | hi people |
| 18:48.30 | citescape | might anyone have a working configuration for outbound faxing using "Fax for Asterisk" ?? |
| 18:48.55 | citescape | I'm really stuck getting it to send outbound faxes and seeing a dialplan that actually works would help me a lot. |
| 18:49.17 | eduardonunesp | sorry by the stupid question but, the option "i" in Dahdi Dial, is only for ensure to use the same SPAN to transfers ? |
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| 18:51.10 | roxdragon | kaldemar, |
| 18:52.21 | roxdragon | when try to call "403" don't work... 403 = sip |
| 18:53.28 | *** join/#asterisk darkspline (~darksplin@ool-182fb966.dyn.optonline.net) |
| 18:55.25 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
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| 19:01.10 | roxdragon | i have a problem.. chan_sip.c:14847 handle_request_invite: Call from '402' to extension '....dyndns.com' rejected because extension not found. |
| 19:04.16 | bdonegan | does anyone know what would cause this? 'X-Asterisk-HangupCause: Bearer capability not available' |
| 19:04.40 | bdonegan | getting this between two soft phone clients |
| 19:08.21 | *** join/#asterisk cashback (~mac@ip68-2-140-46.ph.ph.cox.net) |
| 19:11.49 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 19:12.52 | antiwire | hey, does anyone know how I can setup an asterisk SIP peer to send and receive hook flash? |
| 19:12.59 | antiwire | is that possible with SIP? |
| 19:14.39 | citescape | how can I turn on status events for Fax For Asterisk? |
| 19:19.31 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 19:20.16 | roxdragon | hi |
| 19:21.05 | roxdragon | i have exten => s,1,Dial(SIP/401&SIP/402&SIP/403) |
| 19:21.17 | roxdragon | if an internal call will ring all. How do I change it to ring only the domestic interested? |
| 19:21.26 | Qwell | what? |
| 19:22.01 | *** join/#asterisk eject_ck (~eject_ck@83-218-246-246.dynamic.vega-ua.net) |
| 19:22.19 | eject_ck | Hi guys, what SAY_ should I use to hear 7.9633 |
| 19:22.31 | eject_ck | number says me only "seven" |
| 19:22.36 | eject_ck | say_number |
| 19:23.24 | roxdragon | when try call an internal, ring all internals.. |
| 19:23.39 | roxdragon | how to solve? |
| 19:23.56 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 19:23.58 | Qwell | roxdragon: You really aren't making any sense.. |
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| 19:26.34 | roxdragon | i used google translate |
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| 19:29.33 | munson | any cisco 7942g gurus that could possibly point me in right direction to get these phones enabled in my asterisk/freepbx/freeswitch whatever box. I d/l the SIP fw from cisco and phone is still looking for a P03-8-12-00 file which the fw didn't come with that file |
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| 19:54.56 | benngard | [Mar 17 20:58:25] WARNING[17702]: res_fax_spandsp.c:367 spandsp_log: WARNING T.30 ECM carrier not found <-- any danger with that message, or can u avoid it? |
| 20:05.13 | moos3 | anyone use a qoutcon in real time queues ? |
| 20:05.38 | Qwell | a what? |
| 20:06.12 | moos3 | Qwell quit context in queues at are real time |
| 20:06.21 | moos3 | I have it set but its not working |
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| 20:46.19 | megalomano | hi ,,, can someone explain me if is possible send code SIP 503 to terminate some calls , thanks |
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