| 00:00.02 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:02.10 | drmessano | WIMPy: I would tend to agree.. that so-called expert poster seems to be a career forum poster who gets off spreading his small bucket of knowledge. I dont know why his vague, incorrect statements are of any weight. If you put him on IGNORE and read the rest, it looks like theres a problem with callerID in general and VoLTE is really screwed up |
| 00:03.39 | WIMPy | I agree with the first half. So far I don't really se an issue with VoLTE other than it decides to hand bad caller IDs by dropping them. |
| 00:04.01 | WIMPy | That's something the first switch/server should have done. |
| 00:04.07 | drmessano | If a customer is bitching about it, the correct answer is "This is a widespread, but intermittent problem spread across random calls and random users.. in some places its more prevalent than others. The carriers are more concerned with rolling out VoLTE as a stage 1 than fixing the issues, which is a stage 2 or stage 3 problem" |
| 00:04.36 | drmessano | "Thank you and have a nice day.." |
| 00:04.37 | drmessano | lol |
| 00:05.47 | drmessano | WIMPy if anything I would tend to guess that unvalidated callerIDs are not being allowed to pass, to prevent spoofing. |
| 00:06.15 | drmessano | So if your CID isn't in some database and tied to your account, you're not gonna pass |
| 00:06.23 | WIMPy | That's the way it sould be. |
| 00:06.34 | drmessano | Which is probably more a FIX to the longstanding issue than anything |
| 00:06.42 | WIMPy | But we all know that ITSPs often don't care. |
| 00:07.34 | drmessano | Right.. they err on the side of "Throw any CID in there you want, please dont be illegal and nasty"... So even the customers with proper CIDs are having issues because of the lack of a database cross reference |
| 00:07.51 | WIMPy | Some ITSP will even pass on user imput completely unvalidated as authoritative. |
| 00:08.09 | drmessano | I would imagine some CLECs are the same with SIP and PRIs |
| 00:08.25 | WIMPy | Possible |
| 00:08.58 | drmessano | The CLECs I have deal with on PRIs basically dropped them to SIP and threw them onto a data connection |
| 00:11.47 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 00:13.09 | drmessano | So I guess it's hard to say.. it could be properly dropping invalidated CIDs or just growing pains.. they are throwing up VoLTE sites here faster than anything I have ever seen. At the end of the day, this comes down to having a spine with your customer and telling them the carrier is either dropping CID's for a large subset of calls that don't validate somewhere else, or CID is just broken. Either way, same result |
| 00:14.17 | drmessano | I'm not going to switch from a CLEC or ITSP because they haven't spent the money to implement a validation scheme that wasn't necessary prior |
| 00:15.29 | drmessano | IMHO, CallerID is a joke. It can't be relied on, and is merely a convenience issue for dumb end users that have reason to ignore calls |
| 00:15.47 | WIMPy | It used to be reliable. |
| 00:16.07 | drmessano | Sure, back when you had to rent your phone from Ma Bell :) |
| 00:16.25 | WIMPy | But that was at a time when a lot of other features were reliable and take for granted that don't work or exist any more. |
| 00:16.49 | WIMPy | *taken |
| 00:17.07 | drmessano | But now CallerID is no different than going by IP Address. "WHATTTTTT??? they spoofed their IP address or used a PROXY? ?" |
| 00:47.03 | jeev | WIMPy, mine is a "pri" |
| 00:47.18 | jeev | although it's a pri to a local adtran, then private layer 2 to telepathetic |
| 00:47.59 | WIMPy | Nice. That gives you already two points of failure on your side. |
| 00:48.51 | jeev | WIMPy, it's not my choice |
| 00:49.01 | jeev | i use vz enterprise sip with 11 endpoints |
| 00:49.23 | WIMPy | Who manages the adtran? |
| 00:49.30 | jeev | telepathetic |
| 00:50.09 | WIMPy | Ok, so not your business. |
| 00:50.38 | WIMPy | Then you can only confirm that you send out valid data on your PRI and everyting else is up to your ITSP. |
| 00:51.11 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-waqnnlsbentosgyd) |
| 00:51.37 | jeev | itsp works, whetner it's me or any provider, pri doesn't |
| 00:52.02 | WIMPy | pri doesn't what? |
| 00:53.09 | jeev | pri calls the VoLTE don't pass caller id |
| 00:53.13 | jeev | itsp calls to VoLTE pass caller id |
| 00:53.24 | jeev | although VoLTE stuff with verizon claims caller id (name) doesn't work |
| 00:53.28 | jeev | it works finen when i call from vz |
| 00:54.07 | WIMPy | Then check if it makes sense what you send out on that pri. |
| 00:55.33 | jeev | i can't login to the adtran to dump it however it works to iphone 5's for example, caller id passes |
| 00:56.14 | WIMPy | But you can log in to whatever you have connected to it? |
| 00:56.35 | jeev | the asterisk machine. sure. |
| 00:56.56 | jeev | what should i dump, pri? what's the difference though if it works to non VoLTE enabled devices over the same PRI ? |
| 00:57.40 | WIMPy | The differenc is that others don't seem to care much about bad caller IDs. |
| 00:58.05 | jeev | others? providers? |
| 00:58.52 | WIMPy | Seemingly everything non-LTE. |
| 00:59.40 | jeev | * Advanced Calling 1.0 Requirements: HD Voice requires both parties to use a Verizon HD Voice-enabled phone on Verizonâs 4G LTE network. Video Calling requires HD Voice and a Video Calling-enabled phone. HD Voice and Video calls are billed like regular voice calls. Data usage also applies to Video Calls. The following features/services are incompatible with Advanced Calling 1.0: prepaid |
| 00:59.40 | jeev | service, Caller Name ID and Slacker Radio Tones. Additional incompatible features apply for Business and Government customers. |
| 00:59.53 | jeev | that's verizon, now some are saying the caller name id means caller id, i dont know/ddon't care |
| 01:00.31 | WIMPy | no |
| 01:01.09 | WIMPy | It just means they won't do name lookups for you, wich is something different.. |
| 01:01.18 | jeev | i understand that. |
| 01:01.50 | WIMPy | Then don't come up with things that don't matter. |
| 01:01.52 | jeev | the howardforums post guy said: Hi, I'm the OP... It literally shows the word "Unknown" on the screen. I get no number at all. |
| 01:01.58 | jeev | i'm going based off a post. |
| 01:03.07 | WIMPy | We've been there hours ago. Look at what you send. Then you know if you can configure thing or if you have to blame your provider. |
| 01:03.22 | jeev | i'll blame the provider, i have no issues on my side |
| 01:04.40 | WIMPy | Great. Why din't you say so before asking? |
| 01:04.57 | jeev | i said it quite a few times |
| 01:05.07 | jeev | i said i can call itsp from same pbx |
| 01:05.32 | *** mode/#asterisk [-o mjordan] by ChanServ |
| 01:05.33 | WIMPy | That's not the same. |
| 01:05.46 | *** part/#asterisk mjordan (mjordan@nat/digium/x-bqqrqeefblaszkuh) |
| 01:05.59 | jeev | ok |
| 01:11.20 | *** join/#asterisk mjordan (mjordan@nat/digium/x-swvgujpqladguvxy) |
| 01:11.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:19.27 | *** join/#asterisk CaptainPants (mjordan@nat/digium/x-qsnkruugjgiqibro) |
| 01:20.07 | *** join/#asterisk mjordan (mjordan@nat/digium/x-zljvkcwxakjesaav) |
| 01:20.07 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:04.30 | *** join/#asterisk infobot (ibot@rikers.org) |
| 02:04.30 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 02:49.22 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 02:54.46 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:20.08 | *** join/#asterisk Hsilamot (~Hsilamot@187-163-134-189.static.axtel.net) |
| 03:32.18 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-58-46-12.lsanca.fios.verizon.net) |
| 04:22.22 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 04:26.00 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 04:39.48 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 05:02.58 | *** join/#asterisk Hsilamot (~Hsilamot@187-163-134-189.static.axtel.net) |
| 05:30.18 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 05:46.15 | *** join/#asterisk lawsont (~lawson@23.106.83.86) |
| 05:47.19 | lawsont | Does anyone have a good solution we can deploy to allow us to use our e-mail to send faxes through Elastix? ie, 845102395@domain.com forwards to persons fax machine. Thank you |
| 06:09.08 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 06:25.07 | *** join/#asterisk vader- (~Adium@pool-173-49-160-70.phlapa.fios.verizon.net) |
| 06:29.12 | *** join/#asterisk jiku (~jiku@113.193.187.147) |
| 06:38.24 | *** join/#asterisk Leseratte (~florian@2001:470:70bc::1) |
| 06:47.07 | *** join/#asterisk mokmeister (~quassel@95.45.43.214) |
| 06:54.44 | *** join/#asterisk russellb (~russellb@redhat/russellb) |
| 06:54.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 07:29.43 | *** join/#asterisk jiku (~jiku@113.193.187.147) |
| 07:37.17 | drathir | mornin... |
| 07:38.08 | drathir | im searchin a little tip about more motif where to find for debian... |
| 07:41.22 | drathir | or that drivers are builid inside asterisk? |
| 07:45.31 | *** join/#asterisk gusto (~gusto@2001:4c50:62e:a00:8219:34ff:fecf:17f0) |
| 07:58.11 | Penguin | If you have asterisk installed, check the modules directory for the motif channel driver. |
| 08:01.03 | *** join/#asterisk tuxd00d (~tuxd00d@ip70-162-151-19.ph.ph.cox.net) |
| 08:04.39 | *** join/#asterisk K0HAX (~K0HAX@c-75-72-143-131.hsd1.mn.comcast.net) |
| 08:41.38 | *** join/#asterisk tuxd00d (~tuxd00d@ip70-162-151-19.ph.ph.cox.net) |
| 09:03.47 | *** join/#asterisk chandoo (~chandoo@ool-4a59659f.dyn.optonline.net) |
| 09:06.34 | skyroveRR | In which configuration file can I prevent asterisk from sending an email when a voicemail comes in? I only want to store the voicemails, I don't want to get notified through email. |
| 09:07.30 | WIMPy | The obvious one |
| 09:07.35 | WIMPy | voicemail.conf |
| 09:08.02 | skyroveRR | WIMPy: which option is it? Can't seem to find one in the reference manual.. |
| 09:08.28 | WIMPy | Have you tried to look at the file? |
| 09:08.35 | skyroveRR | Yes. |
| 09:09.19 | skyroveRR | Should I just comment out the serveremail option? |
| 09:10.20 | WIMPy | I'd remove the email-addresses from the mailboxes, but there surely are ways to do it globally. |
| 09:15.16 | skyroveRR | I set sendvoicemail=no; commented the [default] section and the serveremail option. |
| 09:15.18 | *** join/#asterisk ghostlines (~ghostline@i191002.upc-i.chello.nl) |
| 09:16.45 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 09:30.15 | tuxd00d | Apparently Randal Law will be asking questions in the future. http://lists.digium.com/pipermail/asterisk-users/2016-May/thread.html |
| 09:32.32 | *** join/#asterisk Visage (~visage@pdpc/supporter/active/visage) |
| 09:34.58 | skyroveRR | lol 2016. |
| 09:44.00 | *** join/#asterisk pmd` (~ospf@44.251.187.81.in-addr.arpa) |
| 09:49.17 | *** join/#asterisk zamba (marius@flage.org) |
| 09:58.30 | skyroveRR | I'm trying to create a simple dialplan where when I dial '1' in DTMF, the call should go to voicemail, but when I call and dial 1, the call exits after 'good-bye'. I guess I haven't configured the extensions or the voicemail files correctly, can someone please point out the errors? Here's my voicemail file http://tracefl.in/files/voicemail.conf.txt and my extensions.conf file |
| 09:58.32 | skyroveRR | http://tracefl.in/files/extensions.conf.txt The voicemail line is under the 'TestMenu' context. |
| 10:13.18 | *** join/#asterisk Visage (~visage@pdpc/supporter/active/visage) |
| 10:29.09 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:56.57 | ChannelZ | Perhaps your DTMF isn't working |
| 10:57.45 | ChannelZ | the only way for it to play 'goodbye' is to timeout |
| 11:09.27 | skyroveRR | ChannelZ: ok, so should I lower the sensitivity or increase it? |
| 11:10.07 | skyroveRR | Sometimes it works, sometimes it doesn't. |
| 11:10.30 | ChannelZ | what kind of channel is it? |
| 11:10.54 | skyroveRR | ChannelZ: GSM channel. |
| 11:11.22 | skyroveRR | Landline to GSM/GSM to GSM. |
| 11:12.24 | ChannelZ | Not sure if I can help.. DTMF usually doesn't survive mobile codecs very well |
| 11:12.49 | skyroveRR | Oh, I increased the sensitivity, think it might work... |
| 11:16.51 | ChannelZ | Good luck. I must run off to bed |
| 11:26.21 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 11:27.37 | EmleyMoor | I just upgraded the OS on my Asterisk box, and my dahdi configuration is lost. I have an AEX410PE card with 3xFXS and 1xFXO fitted... how do I get it working again? |
| 11:44.41 | *** part/#asterisk rtfm (rtfm@unaffiliated/rtfm) |
| 11:59.36 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:37.33 | Ove_ | Anyone who has dealed with telintel? |
| 12:37.48 | Ove_ | I have a sip account that registers fine if I use for instance sipsak. |
| 12:37.53 | Ove_ | From Asterisk it does not. |
| 12:38.08 | *** join/#asterisk JosephP (~joe@0.102.125.91.dyn.plus.net) |
| 12:38.12 | Ove_ | I don't even get an answer from telintel side when using asterisk. |
| 12:38.31 | Ove_ | What would be the best course of action to debug this? |
| 12:39.13 | WIMPy | Compare what both are sending and find the difference. |
| 12:42.06 | Ove_ | The first this I notice is that the packets going out towards the Internet are using the private ip. Whereas when I do it from another host I get the wan interface of the routers IP address. |
| 12:42.18 | Ove_ | As if it is not getting nated |
| 12:44.05 | Ove_ | Pings get nated from the very same box. |
| 12:46.24 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 12:50.27 | Ove_ | Think I got it to Register now. \o/ |
| 13:03.29 | EmleyMoor | I'm really stuck. I want to get dahdi working on my Debian jessie system |
| 13:12.29 | *** join/#asterisk chandoo (~chandoo@ool-4a59659f.dyn.optonline.net) |
| 13:13.45 | *** join/#asterisk fahmad (~fahmad@unaffiliated/fahmad) |
| 13:13.52 | fahmad | hello |
| 13:24.16 | fahmad | i have issue with Application Page & MeetMe its use too much CPU |
| 13:24.22 | fahmad | can we have any alternative for this ? |
| 13:24.30 | EmleyMoor | Got asterisk working just fine, except that dahdi isn't working at all... |
| 13:30.15 | fahmad | hmm |
| 13:31.50 | EmleyMoor | Anyone familiar with dahdi in Debian? |
| 13:31.58 | fahmad | what is the problem |
| 13:32.02 | fahmad | give more details ... |
| 13:49.57 | EmleyMoor | I'm ending up with no channels when I try to configure the services |
| 13:50.20 | EmleyMoor | dahdi_genconf is saying no spans are defined |
| 13:53.15 | EmleyMoor | Tried applying the instructions from here: http://www.sailpbx.com/mediawiki/index.php/V4_DAHDI - but get nothing in the final setup |
| 13:56.24 | EmleyMoor | dahdi_genconf says: Empty configuration -- no spans (three times) |
| 13:58.19 | EmleyMoor | I'll tell you the output of anything, if it would help... |
| 13:58.36 | EmleyMoor | (so often, there's no output - usually that means "success") |
| 14:04.58 | *** join/#asterisk CeBe (~CeBe@port-92-200-36-58.dynamic.qsc.de) |
| 14:07.29 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 14:12.25 | *** join/#asterisk mlhess (~mlhess@drupal.org/user/102818/view) |
| 14:13.22 | fahmad | EmleyMoor: do you have card connected ? |
| 14:13.30 | fahmad | make sure your have card connected properly |
| 14:13.45 | fahmad | shutdown the computer and check the card by removing power cables !! |
| 14:14.54 | EmleyMoor | Card is showing in lspci. It's there. It worked fine in wheezy. |
| 14:15.14 | EmleyMoor | (I seem to recall it was a total swine to set up, though) |
| 14:18.53 | fahmad | hmm |
| 14:24.25 | *** join/#asterisk skyroveRR (~skyroveRR@unaffiliated/skyroverr) |
| 14:24.51 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-zmgbmbztkbtnmbaz) |
| 14:32.10 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 14:32.11 | *** mode/#asterisk [+o file] by ChanServ |
| 15:03.19 | *** join/#asterisk tuxd00d (~tuxd00d@ip70-162-151-19.ph.ph.cox.net) |
| 15:15.50 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 15:16.51 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 15:18.51 | *** join/#asterisk rtfm (rtfm@unaffiliated/rtfm) |
| 15:33.11 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 15:38.49 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 15:41.35 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 15:41.58 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:42.04 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.127.5) |
| 15:42.13 | voip | morning |
| 15:52.21 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 16:01.33 | *** join/#asterisk skyroveRR (~skyroveRR@unaffiliated/skyroverr) |
| 16:42.04 | *** join/#asterisk gusto (~gusto@2001:4c50:62e:a00:8219:34ff:fecf:17f0) |
| 16:58.11 | drathir | mornin, mornin... |
| 16:58.48 | drathir | wonder if any recomendations of google and similar services? most up2date ones? |
| 16:59.05 | [TK]D-Fender | What "service" are you talking about? |
| 16:59.11 | [TK]D-Fender | Google does HUNDREDS of things |
| 16:59.12 | drathir | configuration guides ofc have on mean... |
| 16:59.18 | [TK]D-Fender | I have an Android phone .... |
| 16:59.27 | [TK]D-Fender | That's devloped by Google |
| 17:00.26 | drathir | best all workin with asterisk... looking for working ones, bc wanna to learn how structures workin... |
| 17:00.52 | [TK]D-Fender | You still haven't confirmed WHAT SERVICE you are talking about |
| 17:01.00 | drathir | slowly learning the dopendencies which one connecting where and similar things like that... |
| 17:01.03 | [TK]D-Fender | You are being very vague |
| 17:01.40 | drathir | google voice and hangouts i guess at start should be good start? |
| 17:02.59 | drathir | honestly little hard for me directly point to things, bc of still learning and discover what is possible what not... |
| 17:03.10 | [TK]D-Fender | Forget Google |
| 17:03.14 | [TK]D-Fender | That's going out the door |
| 17:03.23 | [TK]D-Fender | ~itsp |
| 17:03.23 | 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. |
| 17:03.24 | [TK]D-Fender | ^ |
| 17:04.17 | drathir | thanks a lot, i will chek that too... |
| 17:05.07 | drathir | asterisk looks for now so powerfull, even when dont know so much about it... |
| 17:06.29 | [TK]D-Fender | Asterisk is a telephony toolkit that you can build up for a huge number of different purposes |
| 17:06.31 | drathir | also very like idea have all in one place what similar xmpp doin with transports... |
| 17:06.50 | [TK]D-Fender | I use Asterisk as a jukebox and coffee maker. |
| 17:07.13 | drathir | automation is nice thing too... |
| 17:17.50 | *** join/#asterisk u0m3 (~u0m3@109.96.156.18) |
| 17:20.58 | *** join/#asterisk chandoo (~chandoo@ool-4a59659f.dyn.optonline.net) |
| 17:26.18 | *** join/#asterisk ghostlines (~ghostline@i191002.upc-i.chello.nl) |
| 17:27.32 | *** join/#asterisk ghostlines (~ghostline@i191002.upc-i.chello.nl) |
| 17:43.26 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 17:51.54 | *** join/#asterisk steelbrain (~Anees@182.187.100.110) |
| 17:51.59 | steelbrain | Anyone up? |
| 17:52.39 | steelbrain | I have a scenario where I have a context that does `exten => _X.,1,Wait(999)` then I have a background daemon logged into ami, it redirects that call to a channel where it should be |
| 17:53.11 | steelbrain | The problem is that asterisk becomes kind-of unresponsive, and doesn't even stop when we do `service asterisk stop`, I have to reboot the whole server, it happens after every few calls |
| 17:53.33 | steelbrain | I think, It's because the redirect command doesn't cancel the current wait command. |
| 17:54.01 | steelbrain | Any kind-of help would be appreciated. |
| 17:58.37 | [TK]D-Fender | Show us the failure, and tell us what you're actually running |
| 18:11.15 | steelbrain | [TK]D-Fender: Failure is that asterisk becomes unresponsive. It doesn't stop when we stop it via service and it doesn't accept any new user logins, it just times out on the client app |
| 18:12.38 | steelbrain | [TK]D-Fender: [custom-wait] |
| 18:12.39 | steelbrain | exten => _X.,1,Wait(999) |
| 18:13.11 | [TK]D-Fender | You've already shown taht singular line fo dialplan |
| 18:13.18 | [TK]D-Fender | Repeating it doesn't offer anything more. |
| 18:14.11 | [TK]D-Fender | Showing what the processes CPU usage & status, etc would be more useful, as well as answering the rest of what I asked |
| 18:14.41 | steelbrain | [TK]D-Fender: BTW what does asterisk do when it's waiting on a wait command and we tell it to redirect it to another channel using asterisk manager interface |
| 18:21.08 | steelbrain | brb |
| 18:24.34 | *** join/#asterisk steelbrain (~steelbrai@182.187.100.110) |
| 18:24.41 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 18:25.04 | steelbrain | back |
| 18:42.20 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:47.16 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 19:01.55 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:16.40 | EmleyMoor | I can't get dahdi set up for my AEX410PE card in Debian jessie - I see "Empty configuration -- no spans" three times from dahdi_genconf. Card worked fine in wheezy. Any advice? |
| 19:24.16 | *** join/#asterisk shinydog (~almostwor@unaffiliated/almostworking) |
| 19:28.05 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 19:28.23 | EmleyMoor | Any ideas on my Debian and AEX410 problem? |
| 19:28.45 | [TK]D-Fender | Stop wasting time on genconf and jsut put in the few lines of config you need yourself |
| 19:28.50 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 19:29.11 | EmleyMoor | [TK]D-Fender: Tried restoring my old config from a backup. Still no go |
| 19:29.38 | [TK]D-Fender | Well it won't be "empty". So show us an actual failure, along with dmesg for the card |
| 19:31.39 | EmleyMoor | dmesg at http://paste.debian.net/169137/ |
| 19:32.51 | *** part/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 19:34.46 | EmleyMoor | Looks like it's having trouble with the firmware |
| 19:34.51 | [TK]D-Fender | Let's see a proper manual config and test |
| 19:38.10 | EmleyMoor | Config to follow in just a moment - but not sure about the test |
| 19:39.33 | EmleyMoor | /etc/dahdi/system.conf is 169138 on same pastebin |
| 19:39.41 | [TK]D-Fender | link it |
| 19:40.23 | [TK]D-Fender | ok, so far so good on that. |
| 19:40.42 | [TK]D-Fender | do just provide the link directly from now on though |
| 19:40.48 | [TK]D-Fender | dahdi_cfg -vvvv |
| 19:40.48 | [TK]D-Fender | ^ |
| 19:42.22 | EmleyMoor | http://paste.debian.net/169139/ - dahdi_channels.conf |
| 19:43.42 | EmleyMoor | http://paste.debian.net/169141/ - dahdi_cfg -vvvv output |
| 19:44.53 | EmleyMoor | (must say, some weird stuff seems to have happened!) |
| 19:49.08 | EmleyMoor | I need to go out now - but I'll check back later to see if there's anything new to try. |
| 19:52.54 | [TK]D-Fender | It's telling you that your signaling is wrong there |
| 20:13.36 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 20:20.13 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 20:21.40 | *** join/#asterisk chandoo (~chandoo@ool-4a59659f.dyn.optonline.net) |
| 20:32.40 | *** join/#asterisk cstk421 (~cstk421@173-14-40-58-Michigan.hfc.comcastbusiness.net) |
| 20:40.38 | *** join/#asterisk c|oneman (cloneman@2605:6400:2:fed5:22:0:3b06:3913) |
| 20:48.25 | lawsont | We are trying to find a good way to use our e-mail to send faxes through Elastix. If you have a recommendation that would be appreciated. ie. 3814950133@domain.com forwards to 3814950133 physical fax machine. Thank you |
| 20:48.35 | *** join/#asterisk Hsilamot (~Hsilamot@187-177-21-35.dynamic.axtel.net) |
| 20:52.51 | *** join/#asterisk cstk421_ (~cstk421@24.114.70.43) |
| 21:10.03 | *** join/#asterisk cstk421 (~cstk421@24.114.70.43) |
| 21:14.59 | *** join/#asterisk rogerh (~rogerh@202.37.70.18) |
| 21:16.30 | *** join/#asterisk cstk421 (~cstk421@24.114.70.43) |
| 21:18.49 | *** join/#asterisk cstk421 (~cstk421@173-14-40-58-Michigan.hfc.comcastbusiness.net) |
| 21:20.38 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-votjljtvisdqthkz) |
| 21:22.34 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 21:49.03 | *** join/#asterisk cstk421_ (~cstk421@24.114.70.43) |
| 21:49.51 | *** join/#asterisk cstk421 (~cstk421@173-14-40-58-Michigan.hfc.comcastbusiness.net) |
| 22:05.54 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 22:05.57 | xochilpili | hi all |
| 22:06.17 | xochilpili | is there a way to make asteriskNOW a provisioning server for cisco spa504G phones ? |
| 22:10.30 | *** join/#asterisk zamba (marius@flage.org) |
| 22:16.21 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 22:28.43 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 22:34.08 | EmleyMoor | [TK]D-Fender: Yes, but why? It hasn't been changed since it last wasn't wrong. I suppose it's vaguely possible that channel 1 might have coincidentally failed - in order to help prove this, how can I "skip" it? |
| 22:34.38 | *** join/#asterisk italorossi (~Adium@179.211.186.236) |
| 22:39.07 | EmleyMoor | I've just mried the most obvious way - the error merely occurs on channel 2... |
| 22:39.13 | EmleyMoor | tried* |
| 22:48.11 | *** join/#asterisk ghostlines (~ghostline@i191002.upc-i.chello.nl) |
| 22:50.16 | *** join/#asterisk chandoo (~chandoo@ool-4a59659f.dyn.optonline.net) |
| 23:11.24 | [TK]D-Fender | EmleyMoor, I think you've got your ports mixed up |
| 23:11.43 | [TK]D-Fender | xochilpili, provisioning = provide configs for phoe via some protocol. |
| 23:12.46 | [TK]D-Fender | xochilpili, Clearly you are able to setup http/ftp/tftp, etc as services. It's up to you to set those up and ensure your phone accesses it properly and that you have configs in the right place there |
| 23:22.19 | cryptic | are there any good web-based softphones that interface asterisk? |
| 23:22.40 | cryptic | essentially, I'm looking for a way to place calls from a web site |
| 23:39.08 | *** join/#asterisk muanang (~muanang@203.115.183.18) |
| 23:41.05 | *** join/#asterisk muanang (~muanang@203.115.183.18) |
| 23:56.29 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 23:59.36 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |