| 00:01.22 | navaismo | dimitry7, show us the iax debug on a pastebin |
| 00:01.28 | navaismo | ~pb |
| 00:01.29 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 00:01.53 | navaismo | and the iax.conf for both servers |
| 00:08.11 | f0ner00t | Hey Navaismo do you know 3com routers in #asterisk .. LOl |
| 00:08.13 | f0ner00t | Just playing |
| 00:08.20 | navaismo | hahaha |
| 00:08.30 | navaismo | so what is your issue btw? |
| 00:09.42 | f0ner00t | I got two 3com switches I am trying to connect togther.. But the one doesn't even show a light when its connected to the other. The physical connection is good because we can plug a 601 polycom in it... But i think it might be my ip config on the one router. |
| 00:09.54 | f0ner00t | I'll have to hook into the other router tommorrow and see how its configured |
| 00:10.38 | f0ner00t | probally |
| 00:12.14 | navaismo | i see |
| 00:12.27 | navaismo | if everything fails check the manual :) |
| 00:12.36 | f0ner00t | yea the manuals aren't much help |
| 00:12.37 | f0ner00t | lol |
| 00:13.03 | navaismo | there is no reset tool, web gui or its old for that |
| 00:13.27 | navaismo | alo alo dimitry7 did you managed the pastebin??? |
| 00:14.48 | f0ner00t | LOL I'm on a craft port... |
| 00:14.50 | f0ner00t | right now |
| 00:22.12 | navaismo | guess he was kidding about the help |
| 00:24.06 | *** join/#asterisk freeedrich| (friedrich@perplexa.be) |
| 00:26.34 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.28.157) |
| 00:34.00 | f0ner00t | Goodnight! |
| 00:34.18 | *** part/#asterisk f0ner00t (~jvandyke@69-170-21-20.static-ip.telepacific.net) |
| 00:42.27 | dimitry7 | navaismo, my boss did the final part lol, its ok, thank you!!! |
| 00:42.48 | dimitry7 | see ya guys, good day, afternoon or night :) |
| 00:50.35 | *** join/#asterisk mzb- (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 00:56.16 | ruben231 | navaismo: my issue is if i used small codecs like g729, audo quality gets worst.. |
| 01:00.02 | WIMPy | ruben231: That's the idea. |
| 01:00.53 | *** join/#asterisk Surye (~Surye@li112-98.members.linode.com) |
| 01:01.36 | Surye | Are there no RPMs for Asterisk 11? Or am I looking in the wrong place? I'd like to upgrade my asterisk 11 and http://packages.asterisk.org/centos/5/asterisk-11/x86_64/ has none |
| 01:01.37 | LieutPants | [ASTERISK-11] [Status: Closed] AGI channel_status failure - https://issues.asterisk.org/jira/browse/ASTERISK-11 |
| 01:03.52 | *** join/#asterisk wdoekes (~walter@wjd.osso.nl) |
| 01:04.01 | navaismo | ruben231, do you want quality pay the price, i mean the bandwidth |
| 01:05.14 | navaismo | see you |
| 01:07.38 | ruben231 | <PROTECTED> |
| 01:09.14 | WIMPy | Too late. |
| 01:10.57 | WIMPy | So what do you want? |
| 01:12.39 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:22.55 | igcewieling1 | Whoever created cdr_adaptive should get a prize or something |
| 01:30.46 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:30.46 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:30.47 | ruben231 | WIMPy: what would be best setup somehow to make this work that quality might be preservde coz based on badwith monitring usage on teh datacenter im only in minimal but the user end are having voice quality issue already. |
| 01:35.41 | WIMPy | How many call and What's the bandwidth? |
| 02:05.18 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 02:09.20 | pigpen | Hi all. I have ran across this before, but I cannot remember the resolution. Asterisk + Polycom phone: when using the "Polycom way" of transferring, the call is not transferred, but ultimately placed on hold after the final "transfer". |
| 02:09.35 | pigpen | The destination number does ring, but when answered, the call is not connected. |
| 02:09.59 | pigpen | after they hang up the errored call, the call is found "on hold" at the origination of the transfer. |
| 02:10.37 | pigpen | digitmap issue? |
| 02:12.29 | igcewieling1 | pigpen: no. likely a user error. Make SURE they are transferring as the polycom docs instruct. We have some phone docs at http://rock.nyigc.net/ |
| 02:13.14 | pigpen | yeah, I even had the "user" (which has managed this pbx for 5 years for 250 users) video the transfer. All was done right. |
| 02:13.36 | pigpen | it sure does sound like user error though |
| 02:14.39 | BludSuckingFiend | hmm |
| 02:14.43 | BludSuckingFiend | Interesting pigpen |
| 02:15.03 | BludSuckingFiend | I just updated some of my IP550s to the latest firmware and I am getting similar behavior |
| 02:15.19 | pigpen | yeah. recently move from asterisk 1.6.12 to 11.2 (wow, big jump) |
| 02:15.23 | BludSuckingFiend | though, I am actually getting an error on the phone display when attempting to complete the transfer |
| 02:15.28 | pigpen | and Polycom 2.2.5 to 3.0.0 |
| 02:15.39 | pigpen | <PROTECTED> |
| 02:15.40 | BludSuckingFiend | and I just upgraded from 1.8.20 to 11 |
| 02:16.25 | pigpen | other than this odditiy, and dealing with user neglect for 5 years across 250 phones, it is going pretty well. |
| 02:16.52 | BludSuckingFiend | I've got about 140, mix of 550s and 501s |
| 02:17.08 | pigpen | wow. 501's |
| 02:17.17 | pigpen | yeah, they have mostly 430's and 601's. |
| 02:17.18 | BludSuckingFiend | I think I just installed 3.0.4 |
| 02:17.28 | BludSuckingFiend | We have maybe 10 501s left |
| 02:17.33 | BludSuckingFiend | the rest are 550s. |
| 02:19.14 | pigpen | yeah the 550's are nice phones. little brother to the 650's |
| 02:19.14 | BludSuckingFiend | 4.0.3.7562 |
| 02:19.19 | BludSuckingFiend | actually, not 3.0.5 |
| 02:19.40 | BludSuckingFiend | they are definitely complex to configure |
| 02:19.43 | BludSuckingFiend | so many options... |
| 02:20.05 | pigpen | yeah. |
| 02:20.07 | BludSuckingFiend | Way more phone than my users need |
| 02:20.14 | pigpen | I have not gone to the 4.x yet. |
| 02:20.43 | BludSuckingFiend | most of my 550s are on 3.2.2 |
| 02:20.55 | BludSuckingFiend | I've upgraded 3 of them as a test group |
| 02:21.01 | BludSuckingFiend | and then ran into my tranfer problen |
| 02:21.03 | BludSuckingFiend | problem* |
| 02:21.42 | BludSuckingFiend | still doing testing and comparing with the older firmware to determine what the problem is |
| 02:22.19 | pigpen | ah, yeah, on the 550's running 3.2.2 |
| 02:23.57 | BludSuckingFiend | I am thinking it is probably a configuration difference |
| 02:24.20 | BludSuckingFiend | since I had to replace the config files when upgrading... so going to try the old configs with the new firmware |
| 02:24.44 | pigpen | yeah, I thought 4 was way different. |
| 02:25.13 | BludSuckingFiend | quite a bit |
| 02:25.29 | BludSuckingFiend | The website on the 550s is totally reworked too |
| 02:25.37 | BludSuckingFiend | opens up a ton more options |
| 02:25.57 | pigpen | oh, my transfer issue is defiantly a phone issue. Using the ## transfer works fine. |
| 02:26.25 | BludSuckingFiend | Yeah, my pre-4.0.3 firmware 550s still transfer ok too |
| 02:26.33 | [TK]D-Fender | 550 = waste |
| 02:26.49 | [TK]D-Fender | costs too mcuh for just a slightly bigger screen than the 450, etc. |
| 02:26.55 | [TK]D-Fender | N o expansion like the 650. |
| 02:27.03 | [TK]D-Fender | Stuck in the middle sad wastes |
| 02:27.13 | [TK]D-Fender | Checkout time, heading home (WAY too late) |
| 02:27.17 | BludSuckingFiend | price for standardization |
| 02:27.43 | pigpen | night TK, good to see you are still alive (I think I am?) |
| 02:28.08 | pigpen | maybe I am a bot and just haven't figured it out yet. |
| 02:35.26 | igcewieling1 | maybe you are a butterfly dreaming you are human? |
| 02:36.29 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 02:43.30 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 02:45.22 | igcewieling1 | we are turning up a customer with 276 channels tomorrow |
| 02:52.49 | andross | you have a customer that needs to be able to make 276 simultaneous calls? |
| 02:54.21 | BludSuckingFiend | that on an asterisk system? |
| 02:56.35 | *** join/#asterisk raden (~Jon@24-240-51-238.dhcp.stpt.wi.charter.com) |
| 02:56.40 | raden | Katty, HUGZZZZZZZZZZZZZZZ |
| 02:56.45 | raden | anyone use asterisk GUI ? |
| 03:01.59 | igcewieling1 | andross: around that yes. |
| 03:02.07 | igcewieling1 | they claim 1 million calls per month |
| 03:02.27 | igcewieling1 | we have a small cluster of asterisk boxes |
| 03:03.48 | BludSuckingFiend | not bad for an asterisk system |
| 03:04.15 | BludSuckingFiend | I've seen a few Avaya PBX/ACDs that handle > 400,000 calls/day though |
| 03:05.52 | andross | cool |
| 03:06.02 | *** join/#asterisk cryptic (~cryptic@pool-71-125-31-38.nycmny.fios.verizon.net) |
| 03:06.14 | andross | i used to work for an ISP supporting hosted asterisk |
| 03:06.31 | andross | we did a few million calls per month |
| 03:07.09 | andross | although that was across all customers we didnt really have any that needed more than a few dozen lines per |
| 03:09.46 | BludSuckingFiend | I was with the 2nd largest outsourced call center company in the world.... pretty impressive phone systems, but terrible company |
| 03:10.07 | igcewieling1 | I think we do about 2 mil calls per month, I was told the customer will increase our call volume by about a third |
| 03:13.21 | BludSuckingFiend | calls came in on an OC-12 |
| 03:13.26 | BludSuckingFiend | that's a LOT of angry customers |
| 03:17.24 | andross | hrh |
| 03:17.26 | andross | heh |
| 03:18.04 | andross | we used a few paetec ds3's on old as5400's |
| 03:18.31 | andross | so thats like around 600 calls per ds3 |
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| 03:20.45 | pabelanger | andross: why did you leave? |
| 03:21.14 | andross | new manager fired me to bring in his own people |
| 03:21.48 | andross | staffing was always an issue and i was overworked |
| 03:24.17 | igcewieling1 | andross: happened to me a few years ago, but it was expected and talked about ahead of time |
| 03:36.55 | andross | its always discouraging getting fired |
| 03:37.02 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-xfbeuqdijfzerdxx) |
| 03:37.04 | andross | but im no longer underpaid and overworked |
| 03:37.17 | andross | which is one benefit to working somewhere else now |
| 03:37.18 | andross | haha |
| 03:39.07 | WIMPy | Has anyone tried to make use of DAHDIs HOLD feature? How far is it possible to make use of it from Asterisk? Does it even make sense to try? |
| 03:41.04 | andross | for moh? |
| 03:42.04 | WIMPy | No, for additional calls. |
| 03:48.24 | igcewieling1 | WIMPy: no, only standard hookswitch call waiting and 3-way calling |
| 03:48.50 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:50.43 | WIMPy | Last time I checkt, 3-way didn't work. But I'm more interested in the network side. |
| 03:51.30 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 03:58.39 | igcewieling1 | PRI or analog? |
| 03:59.33 | *** part/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 04:01.31 | WIMPy | The main use would be BRI. |
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| 04:04.53 | WIMPy | Home / normal office use where this is quite an issue. |
| 04:09.45 | igcewieling1 | ah, 2BCT on ISDN |
| 04:10.00 | WIMPy | No, HOLD/CW. |
| 04:12.34 | BludSuckingFiend | Are you talking about where a B channel is freed up while holding? |
| 04:12.43 | WIMPy | Exactely |
| 04:12.52 | BludSuckingFiend | and I'm surprised anyone still uses BRI |
| 04:13.14 | WIMPy | 30% of all phone lines here are BRIs. |
| 04:13.16 | BludSuckingFiend | I've never seen that done, unfortunately |
| 04:13.50 | BludSuckingFiend | I've seen a lot of BRI-base phones... a lot of proprietary DCP phones are based on it |
| 04:13.58 | BludSuckingFiend | just not much trunking done that way |
| 04:14.14 | WIMPy | DCP? |
| 04:14.38 | BludSuckingFiend | Digital phones... well, that's the Avaya Terminology anyway |
| 04:14.43 | WIMPy | Ah, Avaya |
| 04:14.59 | BludSuckingFiend | eOn digital phones are BRI as well |
| 04:15.21 | BludSuckingFiend | the 4-wire dual-phone loops were a pain |
| 04:15.44 | BludSuckingFiend | 2 phones on one loop taking advantage of that ISDN functionality that allows more than one TE on a loop |
| 04:15.49 | drmessano | I am trying to figure out if a fix that is marked as shipping with 11.3.0 is included in the 11.3.0-rc1. Where can I look to see what revision the 11 branch was at when 11.3.0-rc1 was tagged? |
| 04:21.07 | WIMPy | As far as I see it it will require a rather complex dialplan involvon lots of group foo and possibly some AGI to make it easier, but I'm not sure it's possible to get the full functionality of a PBX. |
| 04:21.11 | WIMPy | Or without a PBX. |
| 04:22.51 | igcewieling1 | drmessano: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.3.0-rc1 ? |
| 04:23.25 | drmessano | Is that 2013-01-30 17:46 +0000 [r380452-380521] ? |
| 04:23.33 | drmessano | 380521 then? |
| 04:26.30 | drmessano | This is driving me nuts lol |
| 04:26.58 | drmessano | I think I found it |
| 04:27.00 | WIMPy | Yes, it would be really helpful if the SVN contained version numbers. |
| 04:28.52 | drmessano | My issue is that there is a somewhat trivial SDP fix that seemed to address misbehaving clients that were offering video.. In the ticket, X-Lite 3.0 was the issue for the reporter. I run a number of BRIA iOS endpoints. After months of sub-12-hour crashes of Ast 11, I threw this patch onto a 11.2.1 install and I have been up and running for 48 hours |
| 04:29.08 | drmessano | However, When I try Ast 11.3.0-rc1, I don't get the same result |
| 04:30.26 | drmessano | Crashes isn't accurate.. The SIP stack seems to go deaf, nothing logged, no errors.. just freezes up until Asterisk is restarted. But anyway, kinda excited but can't figure out why 11.3.0-rc1 isn't doing the job |
| 04:30.57 | drmessano | This issue is addressed in the changelog |
| 04:31.14 | drmessano | GAH ?????? *&*%*^% |
| 04:31.17 | drmessano | lol |
| 04:32.25 | drmessano | I wish I could look at a specific revision in SVN and see the commit, because the patch wasn't applied against the ticket, so there's no notation on the ticket itself |
| 04:38.06 | elguero | drmessano: do you have the Asterisk issue number? |
| 04:38.32 | drmessano | I do.. 20908 |
| 04:40.43 | drmessano | I think this is the closest I am going to get ---> https://code.asterisk.org/code/browse/asterisk/branches/11/channels/chan_sip.c?r2=380331&r1=379393 |
| 04:40.58 | elguero | if you look at the issue, there is a tab Subversion... I think you already found it |
| 04:42.28 | drmessano | So if this is revision 380331, and it APPEARS 11.3.0-rc1 includes up to 380521, it should be included |
| 04:42.35 | elguero | This is another way to look at the commit: http://svnview.digium.com/svn/asterisk?view=revision&revision=380331 |
| 04:45.08 | elguero | correct... patch was committed on 01-29-2013, RC1 was tagged on 01-30-2013 |
| 04:50.02 | drmessano | Yep, the dates seemed to indicate it would have been included, but I am satisfied now from the revisions that it is. |
| 04:50.50 | drmessano | Now to figure out if my testing was flawed, because if this patch is indeed preventing this issue, RC1 should work |
| 04:51.48 | drmessano | otherwise I am running 11.2.1-drmessano-fixes-that-sip-going-deaf-thing-1.0 indefinitely :) |
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| 04:53.26 | WIMPy | You can try to remove the patch from -rc1. Then you know exactely what's in there. |
| 04:54.33 | drmessano | Ah, hadn't thought of that |
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| 09:03.47 | Rhomber | I forgot who i was asking about reparking a call, but it's possible |
| 09:04.17 | Rhomber | you need to configure the parkedcalltransfers and parkedcallreparking options in features.conf :) |
| 09:05.12 | micdobro | Are there any issues related to running 1.6.0 branch on a 64bit system? (I know it's old, but I am kind of forced for legacy :( |
| 09:08.36 | wdoekes | micdobro: not that I know of.. I ran 1.6.2 a long while on 64 bit |
| 09:08.44 | kaldemar | micdobro: maybe not related to 64bit, but the branch does have issues. |
| 09:10.36 | micdobro | wdoekes: thanks :) |
| 09:10.45 | micdobro | kaldemar: "does have issues"? |
| 09:11.13 | micdobro | kaldemar: I'd love just to check if we are aware of the same things |
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| 09:13.29 | kaldemar | the branch got EOL in october 2010. no bugs that are found after that are fixed in it. |
| 09:14.11 | micdobro | ok, I am aware of that |
| 09:14.20 | micdobro | I will be upgrading to 1.8 |
| 09:14.37 | micdobro | but before that happens I'd like to get as much of the legacy 1.6.0 stable |
| 09:14.45 | micdobro | (there's a bit of software written around it) |
| 09:15.25 | kaldemar | 26 newest ones here: http://www.asterisk.org/downloads/security-advisories |
| 09:15.56 | kaldemar | specifically to 1.6.0? if you're stuck with 1.6, at least use 1.6.2 (which is dead too). |
| 09:16.18 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 09:17.24 | micdobro | I haven't checked specifically, but 1.6.0 and 1.6.2 are different releases, meaning there might be different syntax etc. |
| 09:17.36 | micdobro | then I prefer to invest my time into going straight to 1.8 |
| 09:18.51 | kaldemar | why not the latest LTS if you have a choice? |
| 09:19.54 | kaldemar | UPGRADE*.txt in source packages give a pretty good collection of possible syntax and behavioral changes. |
| 09:28.12 | micdobro | yes, that sounds like a viable strategy |
| 09:28.33 | micdobro | anyway, the current production box I'll just move to 64bit system |
| 09:29.12 | micdobro | and then simply start looking into a real upgrade |
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| 09:59.39 | Rhomber | i really wish who ever wrote asterisk-java would have had the common decency to make the code extendable |
| 09:59.44 | Rhomber | *extensible |
| 09:59.49 | Rhomber | everything is friggin private |
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| 11:15.33 | grifx | Hello |
| 11:15.45 | grifx | Is it possible to make a phone call on Analog line with asterisk ? |
| 11:16.43 | grifx | I want to use a Raspberry Pi to make a phone call on analog line. I don't know how to do |
| 11:17.09 | Vince-0 | you need an analogue interface with compatible drivers |
| 11:17.11 | kaldemar | sure. with rpi you need an ATA with an FXO port. |
| 11:18.05 | grifx | Do you know a micro ata with a FXO port ? |
| 11:19.18 | kaldemar | ATA as in analog telephone adapter |
| 11:19.57 | grifx | ok |
| 11:20.09 | grifx | a digital phone will be able to call on an analog line ? |
| 11:20.26 | grifx | or an analog phone will be able to call on a digital line ? |
| 11:22.01 | kaldemar | no and no. |
| 11:22.06 | grifx | http://www.patton.com/products/product_detail.asp?id=328 |
| 11:23.01 | kaldemar | that has one FXS, which is for connecting to a phone. you need one with FXO if you want to use a line. |
| 11:23.29 | kaldemar | such as SPA3102 or similar. |
| 11:28.08 | grifx | kaldemar: thanks you so much. :) I'll study that. I'll come back :) |
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| 11:58.48 | as001 | Hello I managed to get sound with webrtc asterisk and sipml5 demo from google chrome. I must use this patch http://sipml5.googlecode.com/svn-history/r169/trunk/asterisk/asterisk_379070.patch I wonder will this patch be available in new Asterisk 11 releases ? Without it I can not hear any sound during call. |
| 12:00.05 | as001 | I followed this manual and it worked: http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html |
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| 12:16.41 | weinerk | Please help with ideas - I have an intermittent problem: |
| 12:16.41 | weinerk | I have qualify=on for a trunk, sending periodic SIP OPTIONS over UDP, both sides - public IPs (no NAT) |
| 12:16.42 | weinerk | All of a sudden - my OPTIONS cant get through to GATEWAY. |
| 12:16.42 | weinerk | Until for example SIP INVITE originates from GATEWAY - that seems to open traffic in both directions again. |
| 12:17.42 | kaldemar | do you have some ALG enabled in the gateway? |
| 12:17.51 | weinerk | what is ALG? |
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| 12:18.45 | kaldemar | application level gateway. a sip aware sw component in the gateway. |
| 12:20.22 | weinerk | I am pretty sure there should not be. |
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| 12:22.18 | weinerk | Everything was stable for months. Problem started happening a week ago. Daily or more frequently. |
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| 12:30.34 | beefcafe | does asterisk have to be in the media path for srtp? |
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| 13:11.16 | teff | anyone know why I would not be able to hang up a call from csipsimple? Audio is working both ways and dtmf is been sent from the extension, but the hangup button does nothing |
| 13:12.36 | kaldemar | teff: look at sip debug when you try to hang up the call. |
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| 13:15.59 | teff | kaldemar, doesnt appear to regsiter anything once the call is in progress |
| 13:17.34 | kaldemar | then nothing comes from the client to asterisk. |
| 13:19.21 | Katty | hello my asterisk does not work at all, plz answer is urgent thx. |
| 13:22.46 | file | Katty, . |
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| 13:23.35 | igcewieling | Katty: This may help http://campradio.us/tmp/first-good-loaf.jpg |
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| 13:59.32 | Rhomber | can someone explain what subscribecontext means in sip.conf? |
| 13:59.38 | Rhomber | after much reading, i'm still lost |
| 14:05.33 | igcewieling | Rhomber: it is the context for subscribe requests, usually "buddy list" or BLF or other extension state monitoring |
| 14:09.09 | Katty | file: ohai |
| 14:09.15 | file | hi |
| 14:09.24 | Katty | igcewieling: omnomnomnom |
| 14:09.30 | Katty | file: how'rechu |
| 14:09.34 | *** join/#asterisk moos3 (~textual@cpe-72-224-215-87.maine.res.rr.com) |
| 14:09.34 | file | Katty, good! |
| 14:10.00 | moos3 | asterisk manager api when making a socket connection to initate a call how do set the SRC field in the cdr ? |
| 14:10.05 | Katty | file: egggcelent. |
| 14:10.30 | igcewieling | moos3: FUNCTIONS act like VARIABLES |
| 14:10.54 | igcewieling | moos3: "core show function CDR" |
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| 14:12.56 | moos3 | doing this via the manager api |
| 14:13.42 | igcewieling | moos3: do you know how to set a channel variable using manager? |
| 14:13.59 | moos3 | no thats what i'm trying to figure out |
| 14:14.15 | igcewieling | once you do, you'll be 90% of the way there. 8-) |
| 14:14.46 | moos3 | I have the caller id but we are trying to track the support person that generated the call |
| 14:15.05 | igcewieling | moos3: http://www.voip-info.org/wiki/view/Asterisk+manager+API |
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| 14:18.17 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:18.20 | moos3 | i dont think you can override a cdr field from a Variable: src=7000 instead of it using the callerid |
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| 14:27.26 | igcewieling | moos3: with cdr_adaptive you can add your own custom fields to the CDR. |
| 14:27.46 | moos3 | from the socket ? |
| 14:27.54 | igcewieling | we insert the account ID, route id, and billing id into our own fields in the CDR. |
| 14:28.04 | igcewieling | moos3: using CDR() yes. |
| 14:28.31 | moos3 | the thing is i dont want to add any more fields, i just dont want socket generated calls to have incorrect src's |
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| 14:48.19 | moos3 | igcewieling dont think you can call a function from a Socket |
| 14:48.29 | moos3 | I blieve only events can be called |
| 14:48.55 | [TK]D-Fender | there are ways |
| 14:48.56 | igcewieling | moos3: functions are considered variables |
| 14:49.27 | moos3 | socket("CDR: src=7000") ? |
| 14:49.43 | igcewieling | If you can use this: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+SetVar then you should be able to set a variable |
| 14:49.55 | igcewieling | moos3: it is CDR(src)=7000 |
| 14:50.03 | moos3 | oh |
| 14:50.15 | igcewieling | JUST like you set a variable or other function |
| 14:50.25 | moos3 | so socket("SetVar: CDR(src)=7000") |
| 14:50.43 | igcewieling | moos3: no. read http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+SetVar |
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| 14:51.00 | igcewieling | you need to specify the channel name, etc. |
| 14:51.16 | moos3 | k |
| 14:59.23 | kaldemar | moos3: you probably want the variable to be set in the originate action. for that there is the Variable header. |
| 14:59.30 | moos3 | yea |
| 14:59.41 | moos3 | i'm trying to figure out how to set that |
| 15:00.33 | kaldemar | Variable: CDR(src)=value |
| 15:01.54 | kaldemar | however, you'll have an issue with that because the src field of CDR() is read-only. |
| 15:02.49 | igcewieling | kaldemar: will it be updated if CALLERID(num) is changed? |
| 15:03.57 | kaldemar | igcewieling: i'm not sure. i'd expect it to. |
| 15:04.30 | igcewieling | kaldemar: I imagine he could set the right callerid to start with in his Originate? |
| 15:04.50 | kaldemar | originate action also has a header for caller id. if that does not work, another field in CDR can be used, such as accountcode, userfield or something custom. |
| 15:05.07 | kaldemar | igcewieling: exactly. |
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| 15:14.29 | pietro | Hello, |
| 15:14.43 | doolph | hi |
| 15:15.17 | pietro | I need to keep some custom header added by my UAs in invite. Is there a way without using SIP_HEADER() and SIPAddHeader() in dialplan ? |
| 15:16.11 | pietro | I know that asterisk isn't a pure proxy, I'm asking if maybe exists some setting that preserve original headers. |
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| 15:22.08 | [TK]D-Fender | pietro, Yes ... an actual proxy. That is all. |
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| 15:29.27 | MrMeek | how can i debug func_odbc calls to see if i'm getting errors from the sql server? |
| 15:31.54 | igcewieling | pietro: Why is SIP_HEADER not acceptable? |
| 15:32.24 | igcewieling | MrMeek: enable debugging on the sql server. Asterisk's ODBC is not very debug friendly as far as I can tell. |
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| 15:35.54 | pietro | igcewieling: because I need to add more check in my dialplan, But isn't a big issue at the moment. |
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| 15:36.27 | pietro | Someone of development can take a look on this issue ? (patch provided) https://issues.asterisk.org/jira/browse/ASTERISK-19883 thanks, |
| 15:36.28 | LieutPants | [ASTERISK-19883] [Status: Reopened] RTP packet with Timestamp=0 on Multicast paging - https://issues.asterisk.org/jira/browse/ASTERISK-19883 |
| 15:37.24 | moos3 | kaldemar: the reason i'm overrideing the caller id right now is because we dont want internal numbers to show to the world when i call them |
| 15:42.36 | MrMeek | thanks igcewieling |
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| 15:47.26 | mjordan | pietro: in general, development discussions are held in #asterisk-dev. I've gone ahead and noted on the issue that there is a patch |
| 15:48.04 | *** join/#asterisk gtTuna (~Tuna@67.149.75.253) |
| 15:48.35 | gtTuna | What would be some things I should look at for users complaining about intermittent outbound audio cutting in/out? |
| 15:48.43 | pietro | mjordan: thanks |
| 15:49.44 | SuperNull | i want to 'normalize' my cdrs so they are always 1<areacode><numberzzz> not just <numberzzz> i tried doing this by setting CDR(DST) but it doesnt like that.. 'readonly' .. ? |
| 15:50.10 | igcewieling | gtTuna: congestion on your internet connection |
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| 15:50.37 | igcewieling | SuperNull: correct. |
| 15:50.39 | *** join/#asterisk _methods (~minotaur@192.81.210.43) |
| 15:51.04 | igcewieling | SuperNull: you can modify your dialplan so that EXTEN is what you want or you can add a custom field to your CDRs |
| 15:51.27 | SuperNull | soo just modify exten instead.. ? |
| 15:53.07 | igcewieling | SuperNull: exten => _NXXNXXXXXX,1,Goto(1${EXTEN}, 1) and handle everything in your exten => _1NXXNXXXXXX |
| 15:53.25 | igcewieling | this does not work with AEL macros 8-( |
| 15:53.34 | SuperNull | looks like it doesnt work with 1.4 either. |
| 15:53.35 | SuperNull | period. |
| 15:53.39 | Qwell | ~upgrade asterisk |
| 15:53.39 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 15:53.41 | Qwell | period. |
| 15:53.43 | SuperNull | in 1.8 it works flawless. |
| 15:53.59 | igcewieling | SuperNull: huh? my method works on 1.4. |
| 15:54.00 | SuperNull | yeah, well if it were that simple dont you suppose i would do that? |
| 15:54.53 | SuperNull | exten => _NXXXXXX,1,Macro(dial_out,1${CALLERID(num):-10:3}${EXTEN},nanpa) |
| 15:54.53 | SuperNull | exten => _NXXNXXXXXX,1,Macro(dial_out,1${EXTEN},nanpa) |
| 15:54.54 | SuperNull | exten => _1NXXNXXXXXX,1,Macro(dial_out,${EXTEN},nanpa) |
| 15:55.00 | SuperNull | is what i use.. works in 1.8 perfect. |
| 15:55.09 | Qwell | So problem solved. Next? |
| 15:56.14 | navaismo | o/ |
| 15:58.15 | igcewieling | SuperNull: you didn't SAY you were using MACROS |
| 15:58.51 | SuperNull | erm. im using macros. ;) didnt know it made a difference. |
| 15:58.53 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 15:59.00 | igcewieling | yes. |
| 15:59.34 | igcewieling | in your macro try exten => s,1,Goto(${MACRO_EXTEN},1) then create exten lines for _1NXXNXXXXXX |
| 16:00.19 | SuperNull | alright.. gonna require some tweaking. |
| 16:00.27 | igcewieling | we don't use this method anymore because we use AEL, but I used this method for many years on 1.4 |
| 16:01.45 | SuperNull | alright let me try it. |
| 16:06.58 | navaismo | anyone has a chance to chek this sip debug http://pastebin.com/FgtQ5ZpH I cant figure out why receive the "408 Request Timeout" response |
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| 16:16.01 | igcewieling | navaismo: only crazy people use IAX |
| 16:16.23 | navaismo | thanks |
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| 16:25.35 | melter | is there a good source of info for people who want to install asterisk and explore what it can do? |
| 16:26.45 | igcewieling | ~book |
| 16:26.45 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:26.50 | igcewieling | there you go |
| 16:27.08 | igcewieling | navaismo: I'm serious. You won't get much help with IAX because almost nobody uses it. |
| 16:27.52 | kaldemar | navaismo: you'd need to look at the ACOM508, since that what sends it to you. |
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| 16:29.58 | navaismo | right but dont know why, the ringing must be acked?? |
| 16:30.04 | tm1000 | file: Is Asterisk 12 compilable? As in would it be useful to start providing feedback now or should we be waiting a while longer |
| 16:30.30 | file | tm1000, it's Asterisk trunk right now - none of the major stuff has really been merged in, still too early |
| 16:30.37 | tm1000 | file: ok |
| 16:34.51 | gtTuna | igcewieling, yeah, on bandwidth, they not even using half of the T1 they have dedicated for voice |
| 16:35.28 | igcewieling | gtTuna: have you verified with traffic graphs. Our customers tell us that all the time until we show them the graphs showing them otherwise. |
| 16:36.06 | navaismo | igcewieling, i was a big fan of iax2 |
| 16:37.37 | gtTuna | igcewieling, yeah, I run MRTG against their router (Cisco 3725), and also don't see any errors on any of the interfaces |
| 16:41.05 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 16:42.07 | as001 | Hello can you help me with this sip debug http://www.pastebin.ca/2332265 . I tried to make test call from Firefox Nightly + webrtc4all and sipml5 demo phone but I get some error. It works good in google chrome. |
| 16:43.08 | as001 | "chan_sip.c:10427 process_sdp: Can't provide secure audio requested in SDP offer" |
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| 16:46.38 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:57.22 | as001 | I noticed this "Found audio description format telephone-event for ID 101" just before warning for SDP offer |
| 17:01.30 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 17:05.08 | as001 | what is "unknown media description format opus for ID 109" ? |
| 17:05.42 | navaismo | codec opus is not supported yet as far i know |
| 17:06.01 | as001 | is it video codec ? |
| 17:06.19 | navaismo | http://www.opus-codec.org/ |
| 17:06.22 | as001 | thanks |
| 17:07.59 | as001 | but why browser want to comunicate with asterisk with opus insted of alaw codec ? |
| 17:08.29 | as001 | why is that 488 not acceptable here ? |
| 17:10.45 | navaismo | you need to check the codecs that the browser an use, and I have noticed that response using webrtc2sip too, but the call come in |
| 17:10.55 | navaismo | your asterisk is patched? |
| 17:12.48 | moos3 | is there away to make a call via socket just flow the normal call path |
| 17:13.45 | igcewieling | moos3: it is AMI not socket. Use a Local/ channel |
| 17:19.04 | moos3 | igcewieling i'm trying to get this https://gist.github.com/moos3/a7393ba038c4fc962aec to over ride the cdr src field with the ext that dailed it |
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| 17:19.36 | igcewieling | why don't you just set the callerid to be correct when you Originate the call? |
| 17:20.12 | igcewieling | moos3: Channel: Local/$sipext@somecontext |
| 17:20.31 | moos3 | k |
| 17:20.36 | igcewieling | moos3: fix your callerid line and it might work |
| 17:20.46 | as001 | yes my Asterisk is patched |
| 17:20.51 | igcewieling | CallerID: NAME <number> |
| 17:20.59 | igcewieling | NOT CallerID: number |
| 17:21.39 | moos3 | i miean it works just in the cdr table the src are all as 7033555200 |
| 17:21.39 | as001 | I patched it according to this howto: http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html |
| 17:21.45 | moos3 | not as 7000 for example |
| 17:22.19 | moos3 | which i'm trying to fix |
| 17:22.26 | as001 | naviasmo patched Asterisk works with google chrome good but not with firefox nightly |
| 17:22.29 | igcewieling | gix the callerid line forst |
| 17:22.34 | igcewieling | s/gix/fix |
| 17:23.24 | igcewieling | moos3: You must really like pain to write your own AMI code instead of using PHPAGI |
| 17:23.57 | moos3 | igcewieling its inherrited code in a massive applicaiton here are work that I have been made to maintain |
| 17:23.58 | moos3 | :D |
| 17:25.03 | as001 | I enabled all codecs in sip.conf and I have avpf=yes in friend config.I can't believe Firefox can't use alaw codec in communication with Asterisk. |
| 17:26.06 | navaismo | you should see the debug also at firefox to see what happen |
| 17:26.31 | as001 | ok |
| 17:28.04 | moos3 | igcewieling https://gist.github.com/moos3/a7393ba038c4fc962aec |
| 17:28.41 | igcewieling | I still don't see < and > in asterisk("CallerID: ".$sipext); |
| 17:30.31 | *** join/#asterisk psilikon (~joel@mail.vicimarketing.com) |
| 17:31.52 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 17:32.39 | *** part/#asterisk Surye (~Surye@li112-98.members.linode.com) |
| 17:33.56 | as001 | naaismo I can't see antything more in Browser console it stucks at "488 Not Acceptable Here". That is what Asterisk is saying and that is the end. |
| 17:35.06 | moos3 | igcewieling refresh |
| 17:38.27 | navaismo | as001, and with firebug |
| 17:38.46 | *** join/#asterisk jimw (~chatzilla@adsl-74-179-108-63.gnv.bellsouth.net) |
| 17:38.49 | as001 | yes I watched that in firebug |
| 17:39.04 | as001 | just sip messages like in sip debug |
| 17:42.49 | jimw | I have * running on ARM (PogoPlug/Arch Linux) from command line, but when I try to get systemd to start it at boot, it fails: http://pastebin.com/x6Niteas -- I suspect something FUBAR in some .conf file or another, but I'm way over my head. |
| 17:42.52 | as001 | when I use firefox and fail I can see that media opus with id 109, when I use chrome and success I see media opus id 111 but I guess firefox should use opus and alaw and should be able to connect to Asterisk by using alaw codec. |
| 17:45.22 | navaismo | so the issue is with firefox |
| 17:47.00 | navaismo | as001, hrmmm, you dont need to install the webrtc4all or that version dont need it? |
| 17:47.13 | as001 | i Installed that webrtc4all |
| 17:49.54 | navaismo | as001, can you change the codecs only to support codecs in asterisk? I havent used the webrtc4all yet |
| 17:52.13 | as001 | I am afraid I don't have that option in sipml5 demo phone but I guess it is ok because it works from chrome. on asterisk I can configure what I want. I can put allow=all but I don't think it will help. I will try |
| 17:53.54 | as001 | stil the same at sip debug. How can I check if my asterisk is using SRTP ? |
| 17:55.06 | as001 | I have module 'res_srtp.so' loaded and encryption=yes in sip.conf |
| 17:55.14 | navaismo | maybe if the srtp module exist or have count in it: module show like srt |
| 17:55.30 | as001 | res_srtp.so Secure RTP (SRTP) 0 |
| 17:57.07 | navaismo | as001, i cant test from here firefox in linux is unsupported yet, |
| 17:57.11 | Kobaz | so |
| 17:57.31 | Kobaz | is it bad if a girl's ex husband who physicall and mentally abused her, is palling around with the girl's mom |
| 17:58.05 | Kobaz | i really don't understand people like that |
| 17:58.09 | as001 | I am testing from windows firefox but can you tell me one thing. During a call from firefox i should se Use Count 1 for res_srtp.so if it use srtp is that right ? |
| 17:59.44 | navaismo | as001, im not sure, but based on when you use trabscoding the count on codecs show 1(or number with channels active) ill choose yes XD |
| 18:00.50 | as001 | because during call when I do show modules like srtp i can see res_rtp_asterisk.so changed to 1 which if I am correct means I am using rtp not srtp for call and that is exactly what I get in debug. Why is that when I have encryption=yes in friend conf ? Do I need to do some more configuration ? |
| 18:03.17 | as001 | during successful call from chrome both rtp and srtp are 1. |
| 18:03.24 | navaismo | hints are here baout that--> https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support |
| 18:04.12 | *** join/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 18:04.40 | as001 | ok I will check that I guess it is firefox issue... Thanks |
| 18:05.02 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 18:06.40 | as001 | I followed that page already i have all libraries and what Joshua said on that page |
| 18:08.49 | navaismo | yup firefox still nightly |
| 18:09.44 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 18:18.52 | *** join/#asterisk Spengler1 (~spengler@dns1.aacc.edu) |
| 18:18.56 | *** join/#asterisk gusto (~gusto@ppp-93-104-69-167.dynamic.mnet-online.de) |
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| 18:25.49 | *** join/#asterisk fireman_biff (~biff@65.48.133.103) |
| 18:26.26 | fubada | has anyone ever worked with FlexySIP? |
| 18:26.35 | fubada | FlexiSIP |
| 18:28.56 | *** part/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 18:31.21 | fireman_biff | does the term "clocksource" when talking about PRIs relate in anyway to the system time? My system time keeps going off and I'm wondering if there could be a connection. (its not a VM) |
| 18:31.38 | Qwell | fireman_biff: no |
| 18:31.40 | WIMPy | no |
| 18:31.43 | fireman_biff | cool, thanks |
| 18:32.11 | Qwell | Install an ntpd, so it stays tight. |
| 18:33.01 | fireman_biff | I have ntpd installed and running but its still messing up, and loosing a second every 15 seconds |
| 18:33.11 | Qwell | yikes |
| 18:33.14 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
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| 18:41.50 | *** join/#asterisk picard276 (~chatzilla@cpe-67-255-1-14.twcny.res.rr.com) |
| 18:41.57 | picard276 | hey anyone familiar with PJ_SIP |
| 18:42.06 | picard276 | or know a channel to ask pjsip questions? |
| 18:42.22 | *** join/#asterisk marksaitis (~marksaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 18:45.29 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 18:51.01 | *** join/#asterisk poseidon (~joe@vps6967.inmotionhosting.com) |
| 18:51.24 | poseidon | Anyone know of a php api to work with asterisk ami? |
| 18:51.28 | poseidon | *php library |
| 18:56.04 | navaismo | phapagi |
| 18:56.42 | poseidon | navaismo: do you know if it works over the tcp or http? |
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| 19:00.23 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:04.01 | navaismo | you can check it at http://phpagi.sourceforge.net/ |
| 19:06.49 | nixhr | fireman_biff: is it maybe a virtualbox machine? |
| 19:07.10 | fireman_biff | nixhr: no, its not virtualized |
| 19:07.18 | fireman_biff | at this point I'm thinking it might be the cmos battery |
| 19:07.24 | fireman_biff | or some hardware issue |
| 19:08.36 | WIMPy | The RTC is powered by the PSU, usually even if the machine is switched off. |
| 19:09.05 | WIMPy | And the RTC is only read at boot time. After that other timers take over. |
| 19:09.26 | WIMPy | See hwclock. |
| 19:09.50 | nixhr | fireman_biff: strange |
| 19:10.01 | fireman_biff | WIMPy: I was under the impression that the cmos battery only comes into play when the system has no power, but others were telling me differently... so you'd rule out the cmos battery as the cause then? |
| 19:10.19 | WIMPy | yes |
| 19:10.58 | nixhr | fireman_biff: what does the ntpd log say? |
| 19:11.01 | fireman_biff | hmm... 'hwclock --show' shows the wrong time |
| 19:12.17 | WIMPy | As wrong as date ore more wrong? |
| 19:12.24 | WIMPy | -e |
| 19:13.11 | fireman_biff | date is right, but time is off... `date` = "15:11:50" `hwclock --show` = "12:49:09" |
| 19:13.18 | fireman_biff | so even the timezone wouldn't explain that difference |
| 19:13.55 | fireman_biff | so should hwclock --systohc fix that? |
| 19:14.24 | WIMPy | yes |
| 19:14.36 | WIMPy | And hopefilly the RTC should stay correct. |
| 19:14.53 | fireman_biff | any chance running that could interfere with the phones, PRI, etc? |
| 19:15.36 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:15.37 | robert_ | so for some reason, CHANNEL[secure_bridge_signaling] is coming up 0, even for so-called "secure" calls. |
| 19:16.17 | WIMPy | No, but it might have a cause that affects other timing related things as well. |
| 19:18.43 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 19:18.45 | WIMPy | Have you checked voltages? |
| 19:18.58 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 19:19.28 | fireman_biff | haven't checked anything physically yet... you mean the voltages coming out of the PSU? |
| 19:19.38 | WIMPy | yes |
| 19:20.14 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 19:20.28 | fireman_biff | I'll have to do that when I can schedule down time... for today I'm running ntpdate manually via cron and everything is working |
| 19:20.44 | picard276 | is the format for uri registration |
| 19:20.49 | picard276 | sip:user:pass@ip |
| 19:20.53 | WIMPy | sensors |
| 19:21.02 | picard276 | because i try that and it will ont register? |
| 19:21.19 | fireman_biff | I'm not familiar with that... what should I search for? |
| 19:21.34 | WIMPy | lm_sensors |
| 19:22.36 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:27.07 | picard276 | anyone? |
| 19:28.03 | fireman_biff | WIMPy: sensors-detect doesn't detect any sensors |
| 19:28.38 | WIMPy | You might be missing the the right drivers then. |
| 19:29.17 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 19:29.43 | WIMPy | So either you could try your own kernel with everything enabled or wait til you can get there and look in the BIOS. |
| 19:32.37 | fireman_biff | Think I'll check the BIOS tonight, or use a PSU tester if the BIOS doesn't tell me enough |
| 19:32.41 | fireman_biff | thanks for your help |
| 19:34.20 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
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| 19:48.54 | *** part/#asterisk fireman_biff (~biff@65.48.133.103) |
| 19:48.59 | robert_ | oh, nevermind. |
| 19:49.50 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 20:02.56 | *** join/#asterisk areski (~areski@80.174.255.7.dyn.user.ono.com) |
| 20:06.02 | Kobaz | I'm getting this when people are being put on hold: [2013-03-14 16:00:24] NOTICE[12417]: chan_sip.c:25939 check_rtp_timeout: Disconnecting call 'SIP/420-00000f7a' for lack of RTP activity in 61 seconds |
| 20:06.19 | navaismo | change your rtptimeout setting |
| 20:06.32 | Kobaz | yeah but |
| 20:06.35 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 20:06.43 | Kobaz | that's not a proper fix |
| 20:06.47 | Kobaz | musiconhold is generating rtp |
| 20:06.56 | Kobaz | it shoukdn't trigger the timeout in the first place |
| 20:07.19 | robert_ | is there a way I can enable encryption=yes and transport=tls for ALL "users"? |
| 20:09.28 | malcolmd | put it in a template and then add that template to all of your users |
| 20:11.44 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 20:15.23 | robert_ | hm, I can do that in sip.conf? |
| 20:16.49 | navaismo | yep |
| 20:17.03 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 20:17.06 | robert_ | oh, I see. |
| 20:17.18 | robert_ | so I can have like [employee](!) |
| 20:18.49 | *** join/#asterisk gerhard7 (~gerhard7@195-241-233-43.ip.telfort.nl) |
| 20:18.51 | pabelanger | Kobaz, transmitsilence asterisk.conf? |
| 20:18.56 | pabelanger | or what ever the setting is |
| 20:19.56 | robert_ | and then "inherit" all employees from that? |
| 20:21.20 | robert_ | sweet. |
| 20:25.40 | malcolmd | yup… if you have [employee](!) you've declared a template called employee. you can put stuff in it. then, for some other thing that wants to inherit the template, you'd do [bob](employee) and you can inherit multiple templates like [bob](employee,mainoffice) |
| 20:32.12 | ChannelZ | I like [uncle](!) and [bob](uncle) better |
| 20:34.37 | robert_ | :p |
| 20:37.07 | drmessano | What about [uncle](!) and [bob](uncle,dad) ? |
| 20:37.19 | drmessano | (Only for below the mason/dixon line) |
| 20:37.50 | robert_ | lmao |
| 20:38.41 | robert_ | yay, done. |
| 20:43.07 | robert_ | also |
| 20:43.15 | robert_ | it doesn't hang up when the caller hangs up |
| 20:43.16 | robert_ | o.O |
| 20:49.28 | *** join/#asterisk blee (~blee@rrcs-71-43-69-162.se.biz.rr.com) |
| 21:00.53 | *** join/#asterisk blee (~blee@rrcs-71-43-69-162.se.biz.rr.com) |
| 21:08.14 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-159-37.user.veloxzone.com.br) |
| 21:09.20 | *** join/#asterisk blee (~blee@rrcs-71-43-69-162.se.biz.rr.com) |
| 21:12.32 | anonymouz666 | if you mixmonitor (without bridge option) and then you dial(dahdi) to somewhere and starting the early audio and after an answered condition - if you insert a playback(beeperr) into a macro (dial M option) on called channel, you can't listen the beep in the recording (mixmonitor). why is that? |
| 21:14.25 | anonymouz666 | in resume what I saw was if i insert an audio after party B answered, this audio wouldn't be on recording |
| 21:14.51 | navaismo | the channels are bridged at all? |
| 21:14.59 | navaismo | works without the b option? |
| 21:15.15 | anonymouz666 | didn't try with b option. it is bridged. |
| 21:15.23 | anonymouz666 | there is answer. |
| 21:16.53 | anonymouz666 | what I am trying to acomplish is simple, when I dial out the mobile operators starting the mailbox at different points and that makes AMD confused. I just want to mark with a beep in the recording and see when the most of calls are hitting. |
| 21:17.15 | anonymouz666 | to know the right time of answering machines |
| 21:17.30 | anonymouz666 | and tweak amd |
| 21:18.16 | anonymouz666 | mark with a beep when the telco send the ANSWER to start to bill |
| 21:20.24 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:20.24 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:21.59 | navaismo | hmm i have no idea |
| 21:24.17 | anonymouz666 | i dont know if i was clear enough to explain what i am trying to do |
| 21:25.24 | navaismo | always lost in trasnlation |
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| 21:33.02 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xudporutogjpmqlk) |
| 21:34.22 | *** join/#asterisk epaphus (~user1@108.174.50.29) |
| 21:35.19 | epaphus | Hello all.. Iam about to hit my head on the wall. I have an extension that isnt registering. When i enable sip debug i see the server responding with Unauthorized... yet i checked the password in sip.conf for that extension and iam sending the correct one.. as well as the userid... |
| 21:35.34 | epaphus | How ocould i obtain more info? |
| 21:35.53 | epaphus | This is my sip debug http://pastebin.ca/2332436 |
| 21:40.37 | kaldemar | epaphus: your device is not even trying to authenticate. |
| 21:41.48 | epaphus | hmm this is a Sipura.. why could that be kaldemar? |
| 21:42.31 | kaldemar | no idea. but it's the device you should be looking at. |
| 21:42.50 | epaphus | ill reset it :) |
| 21:44.25 | epaphus | it was the nat option on the device, it was set to off... it had to be on.. the weird thing is that i didnt change that hmm |
| 21:44.34 | epaphus | tnx |
| 21:45.26 | epaphus | kaldemar, what clue did you have in that debug that it didnt even try to auth? curious\ |
| 21:59.25 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 22:17.13 | kaldemar | epaphus: it was not responding to the "unauthorized" with a new invite with credentials as it should have. |
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| 22:23.11 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 23:43.05 | darkdrgn2k | hi all, is there anyway to prevent asterisk from sending out TRYING? |
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