| 00:08.00 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 00:14.02 | *** join/#asterisk Penguin (~xwQ5kwYl6@20264.odci.gov.united-states.rltk.us) |
| 00:30.03 | gusto | hey Penguin |
| 00:30.11 | *** join/#asterisk rhamnett (~rick@97e03ac6.skybroadband.com) |
| 00:30.25 | Penguin | hi |
| 00:30.47 | gusto | you have your irc client configured wrongly now ... ##freebsd is no longer ... now you have to join #freebsd |
| 00:30.52 | gusto | i have to adjust that as well |
| 00:31.29 | gusto | today i am done .. i configured a pile of ARM devices with BSD's |
| 00:31.57 | gusto | and i am listening to internet radio over IPv6!!! |
| 00:33.12 | newtonr | Any one know of a SIP soft phone that supports TLS, SRTP and can do call transfer or forwards ? (with a 302 Moved Temporarily) ? |
| 00:33.47 | gusto | linphone is far with supporting this, but i am not sure if it does it all |
| 00:34.27 | gusto | TLS and SRTP should not be a problem but that transfer ?? no idea |
| 00:34.50 | newtonr | gusto, they say they support it on http://www.linphone.org/eng/features/ , thanks. I had not tried Linphone in a long while |
| 00:35.09 | gusto | http://www.linphone.org/eng/features/ |
| 00:35.33 | gusto | SIP/UDP, SIP/TCP, SIP/TLS |
| 00:35.34 | gusto | Supports SRTP and zRTP (encryption of voice and video) |
| 00:36.05 | gusto | i love linphone for its debug console |
| 00:36.32 | gusto | that one is comparable to asterisk sib debug |
| 00:36.40 | gusto | sip debug |
| 00:38.28 | newtonr | neat |
| 00:39.43 | newtonr | gusto, thank you |
| 00:40.32 | newtonr | It is not obvious how to configure the client for TLS SRTP, hrm |
| 00:40.35 | newtonr | a googling I go |
| 00:48.53 | *** join/#asterisk sectech (~sectech@stjhnbsu0ww-142134150058.dhcp-dynamic.FibreOP.nb.bellaliant.net) |
| 00:49.55 | gusto | i love my serial usb adapter ... without it i could not do anything |
| 00:50.52 | newtonr | I love USB in general, so much less fiddling with adapters than the old days |
| 00:51.29 | gusto | no, i mean that adapter that i can connect to serial console |
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| 00:52.03 | gusto | it is usb, but however, without it you can not connect to say beaglebone or raspberrypi to see what is going on |
| 00:52.52 | gusto | one could use it also on TV sets or routers that have rs232 compatible interface |
| 00:52.58 | newtonr | I know what you mean. I used to use USB to Serial adapters quite a bit. |
| 00:53.24 | gusto | yes, i have a custom one with pins for 5V and 3.3V |
| 00:53.25 | newtonr | I was saying that since USB has become more popular, in general we have less adapters to mess with. |
| 00:53.34 | newtonr | I also wish we had these http://www.universalnetworkcable.com/ back in the day |
| 00:53.42 | gusto | yes, but USB is only the vehicle |
| 00:54.25 | gusto | rs232 and jtag are also standards |
| 00:54.49 | newtonr | righto |
| 00:55.16 | gusto | today i managed to get netbsd, freebsd and openbsd to work on raspberrypi and beagle bone black |
| 00:55.21 | gusto | everything works |
| 00:55.38 | newtonr | nice |
| 00:55.41 | gusto | yes |
| 00:55.56 | gusto | netbsd and freebsd on raspberry and openbsd on beagle bone black |
| 00:57.08 | gusto | the new current version of openbsd (5.5) has edma support that makes I/O faster. does about 1MB/s to microSD ... not that challenging, on linux (PC) through a card reader it does 2MB/s or more, but this is acceptable as well, most importantly it is faster than without DMA |
| 00:57.52 | gusto | and it now also knows the internal storage .. .but i am not using it, because that is only 2GB big and i have a 8 GB mSD card in, so i use that one |
| 00:58.25 | gusto | sadly there are no precompiled asterisk packages yet for ARM |
| 00:58.39 | gusto | packages generally yes, but no asterisk |
| 00:59.12 | newtonr | interesting |
| 01:00.59 | gusto | but you know what would be really cool? with linux you can already do it, but when there will be GPIO support for freebsd (maybe it already is) one could buy that DCF77 antenna and have a precise clock for asterisk on it |
| 01:01.29 | gusto | DCF77 is a time signal here in europe ... on 77khz |
| 01:02.11 | newtonr | Oh neat. Is that "atomic clock" ? I've never looked into that |
| 01:02.16 | gusto | yes |
| 01:02.41 | gusto | there are time signals all over the world, DCF is in germany, but there are also some in great britain, USA, japan and russia |
| 01:02.58 | gusto | there is a list on wikipedia |
| 01:03.32 | newtonr | Someone linked me to a wrist watch with atomic clock signal reception the other day, it looked crazy huge. |
| 01:03.55 | gusto | thats maybe because of the low frequency antenna |
| 01:04.22 | gusto | on low frequencies you have huge vawes ;-) |
| 01:04.29 | newtonr | gusto, http://gizmodo.com/this-atomic-clock-wristwatch-only-loses-a-second-every-1433827555 |
| 01:05.05 | newtonr | Oh that one operates differently |
| 01:05.19 | newtonr | "Appropriately called the Cesium 133, the watch is powered by a $1,500 chip called the SA.45s CSAC (short for chip scale atomic clock) that contains everything you need to keep perfect atomic time, including cesium gas, a laser, and a photodiode detector. " |
| 01:05.29 | gusto | ah no, it has an atomic clock included |
| 01:05.47 | gusto | well, if you are not afraid of radiation than its the way to go |
| 01:05.59 | gusto | for ppl who live near fukhusima it makes no difference |
| 01:06.33 | newtonr | I have a plain ol' automatic watch that dpeloquin conned me into buying |
| 01:06.56 | gusto | but for normal people the dvb-t time signal would be good enough |
| 01:07.36 | gusto | or time signal from DAB+ radio ... that would be a good add-on to raspberrypi |
| 01:08.17 | gusto | in times when i had no internet connection, i used the dvb-s signal to adjust my clock inside computer |
| 01:08.34 | gusto | dvbdate is the application you can use |
| 01:09.30 | gusto | it is not as good as NTP but it theoretically could be even better, it is more a problem of implementation, but it gives results quite close |
| 01:09.58 | gusto | maybe it got better in the meantime, i havent tried it for a long time, since i have now internet connection everywhere |
| 01:11.55 | gusto | GSM/UMTS have to have time signals as well, it is broken in my nokia cell phone but that would be also one way how to recieve time |
| 01:11.58 | newtonr | You are in Germany? |
| 01:12.01 | gusto | yes |
| 01:12.06 | gusto | i am in nuernberg now |
| 01:12.17 | newtonr | cool |
| 01:12.21 | gusto | why? |
| 01:12.27 | gusto | where are you? |
| 01:12.41 | newtonr | Was just curious, as you mentioned Europe earlier |
| 01:12.51 | newtonr | Oh you did say that the DCF is in Germany |
| 01:13.04 | gusto | there is nothing going on around here except maybe Ukraine |
| 01:13.05 | newtonr | I'm in Huntsville, AL, here in the Digium offices |
| 01:13.22 | gusto | so USA |
| 01:13.32 | newtonr | Yeah! I forgot the country :D |
| 01:13.44 | gusto | americans act like there would be no other country |
| 01:14.02 | newtonr | Well, it is more like I act like Alabama is my country |
| 01:14.14 | newtonr | it is a big place |
| 01:14.16 | gusto | that may be because they are almost alone up there on the north american island |
| 01:14.48 | gusto | with a fence around to keep the mexicans out :-D |
| 01:15.30 | gusto | we do not need fences around here, we have mediterrian see |
| 01:15.32 | gusto | sea |
| 01:16.15 | gusto | and since the EU is broke no one wants to come in anyway |
| 01:17.41 | gusto | many hoped for the Ukrainians but they are burning tires now instea |
| 01:17.44 | gusto | instead |
| 01:18.07 | newtonr | Yeah that is a tough situation for them |
| 01:18.27 | newtonr | I hope it doesn't turn into a civil war |
| 01:18.43 | newtonr | well, I'm out of here. It is almost sleep time! |
| 01:18.44 | gusto | it is far from being a civil war |
| 01:19.52 | gusto | the interesting thing is what the mainstream media does not say and that is that the protests in ukraine are not about EU vs antiEU any more but are more against their own corrupt elite |
| 01:20.36 | *** join/#asterisk tristero (~al.f.zero@unaffiliated/transfinite) |
| 01:20.40 | gusto | a situation very much like in USA where ppl have enough of these clowns |
| 01:21.17 | gusto | government shutdowns and things like that |
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| 02:23.21 | smellis | anyone have any guidance on polycom and asterisk 11 throwing Username/auth name mismatch |
| 02:23.30 | smellis | on 1.6 the same config was working just fine |
| 02:29.46 | gusto | you were using 1.6? that's old! |
| 02:30.24 | gusto | what is that guidance on polycom? is it a telephone or what? |
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| 02:32.22 | smellis | yeah an IP320 configured as a peer in sip.conf, getting Username/auth name mismatch on the console |
| 02:32.59 | smellis | authname and password are good, if I purposefully put in the wrong password, I get wrong password in the console |
| 02:33.00 | gusto | yes |
| 02:33.23 | gusto | when it is a peer he does not check the password |
| 02:33.33 | gusto | did you try to make him type=friend? |
| 02:33.45 | smellis | oops, that's what I meant |
| 02:33.47 | smellis | friend |
| 02:34.07 | gusto | you can also use it without password and username |
| 02:34.12 | smellis | sip show peers has me saying peers, but in sip.conf its a friend, so user and peer |
| 02:34.42 | gusto | just remove any mention of username and password on both sides and confiture it as type=peer with matching IP and insecure=invite |
| 02:35.07 | gusto | when it is configured as friend you also have it in sip show users |
| 02:36.00 | smellis | I want to require a secret to register, so that doesn't seem like a good option |
| 02:36.19 | gusto | however |
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| 02:37.21 | gusto | maybe it does not register from the right port, did you try insecure=port? |
| 02:37.52 | Tecy | under [inter] I created an extension 123 with Wait(3600), how would I call this number in a Dial() command? |
| 02:38.19 | smellis | Dial(Local/123@context) |
| 02:44.36 | Tecy | smellis: hmm, can't get that working: http://pastebin.com/BB2kbCw1 |
| 02:46.08 | smellis | Tecy: not sure you can do that, what are your trying to accomplish? |
| 02:46.41 | smellis | I think Dial needs something that progresses |
| 02:48.05 | Tecy | smellis: my * server is in a datacenter, but my internet sometimes doesn't work. In that case my phone (SIP/555) cannot connect to * and the caller goes to VoiceMail directly. I want it to ring for 20 seconds even though my phone is not reachable |
| 02:48.39 | Penguin | That wouldn't be too hard to write dial plan for. |
| 02:49.02 | Tecy | smellis: actually when I call to 123 directly it keeps ringing for 3600 seconds |
| 02:49.03 | Penguin | I would check to see if the phone is available. If it is not, I would play ringing sound for 20 seconds before going to voice mail. |
| 02:49.09 | smellis | yeah, maybe make a queue that lasts for 20 seconds |
| 02:49.41 | smellis | I guess the idea is that, if your phone comes up within that 20 sec, you'll grab the call? |
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| 02:49.48 | Tecy | Penguin: but then it would answer the phonecall and play an audiofile? |
| 02:50.07 | Penguin | It's your dial plan, so you can write it any way you want. |
| 02:50.24 | Penguin | I think it's weird to provide ringing sound for 20 seconds when the phone is not there. |
| 02:50.47 | smellis | I think using/abusing a queue is the way to go there, because you still have an opportunity to answer the call |
| 02:50.50 | Tecy | smellis: well that's not the most important thing. My internet connection works 99,9% of time. It's just weird for a caller to go to voicemail directly when they call a landline |
| 02:50.53 | Penguin | I would want to call to go right to "unavailable" voice mail if the phone is gone. |
| 02:51.32 | Penguin | smellis: If the phone appears part of the way through the cycle, the phone will not be able to take the current call. |
| 02:51.53 | Penguin | tecy: If the phone is not available, that's what should happen. Right to voicemail. |
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| 02:52.25 | Tecy | Penguin: well in my country it's not usual for a landline number to go to voicemail without ringing |
| 02:52.31 | Penguin | If the phone comes back online during that 20 seconds that you're wanting to wait, you're still not going to get the call onto your phone. |
| 02:52.33 | smellis | Penguin: I agree, but he is wanting to simulate a normal land line, which is always available |
| 02:52.49 | mnathani | how complex is it to setup a DID that can act as a voicemail number that saves the voicemail as MP3 and emails it to me? |
| 02:52.59 | smellis | unless the call was put into a queue |
| 02:53.06 | mnathani | kind of like a do it yourself youmail |
| 02:53.17 | Penguin | mnathani: On a scale of 1 to 10, with 10 being the hardest.... 3. |
| 02:53.24 | smellis | haha |
| 02:53.30 | Tecy | lol |
| 02:53.39 | smellis | Use record, the shell out to mutt and send it as an attachment |
| 02:53.46 | smellis | easy breezy |
| 02:53.47 | Penguin | I only elevated it to a 3 because you have to turn it into an mp3 outside of asterisk before emailing it. |
| 02:54.10 | mnathani | I guess I could live with asterisk audio format WAV? |
| 02:54.29 | Penguin | You can record the vm as a wav and then convert it before emailing it. |
| 02:54.54 | Tecy | mnathani: http://www.voip-info.org/wiki/view/Asterisk+Voicemail+as+email+using+sendmail |
| 02:57.05 | Tecy | erm, Penguin / smellis, actually my example was correct. Just forgot to put an ,r to Dial() on my * server :P |
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| 04:13.27 | Gokee2 | Is vmail.cgi still the preferred solution for web based voicemail or is there something better out there? |
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| 04:50.24 | VlaDeMaN_ | watup asterisk people! wondering if anyone can clue me in as to y incoming calls aren't handled correctly. i am unable to understand sip debug output correctly. currently running on asterisk 11.5.1 under freepbx 2.11.0.17. i just swapped out our old asterisk server running freepbx 2.9 and the settings are exactly the same |
| 04:51.05 | VlaDeMaN_ | was lurking in #freepbx but no one is around right now, and we are down here at work because of this |
| 04:58.34 | VlaDeMaN_ | sip output http://pastebin.com/9QQzdYKg |
| 05:03.21 | Penguin | I'll give you a piece of advice for the next time you ask for help. People don't like it when you intentionally obfuscate all the pertinent bits and then ask what's going on. |
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| 05:11.53 | VlaDeMaN_ | i thought i was supposed to do that. here is one unmodified: http://pastebin.com/SqsvQjgh |
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| 05:19.40 | VlaDeMaN_ | created a blank inbound route, works fine. can't decode in the debug where the problem is |
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| 05:24.38 | ChannelZ | Retransmitting.. Retransmitting.. |
| 05:28.02 | VlaDeMaN_ | fucking fuck. in the old system the DID in the incoming route didn't need the +1 but in my 11.5.1 it does. |
| 05:29.48 | WIMPy | That's surely not because of Asterisk version. |
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| 05:30.30 | [TK]D-Fender | INVITE sip:+13109097900@65.91.29.105:5060;transport=udp SIP/2.0 <- Asterisk version didn't make your provider sent the invite like that |
| 05:30.44 | Penguin | Probably some goofy FreePBX nuance. |
| 05:30.45 | VlaDeMaN_ | unfortunately i don't know why. the trunk settings had the + prefix in the old system too. |
| 05:32.11 | VlaDeMaN_ | sorry i don't follow... that's the way the provider sent the invite? |
| 05:33.36 | Penguin | Your side of the equation receives it the way it is sent from the other side. Depending on your configuration, you may be able to tell them how to send it. |
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| 07:57.11 | bchamberlain | morning - evening all... How do I stop asterisk sending any caller ID updates to trunks? I've added canreinvite/rpid_update/directmedia=no in sip.conf general and on trunks which has definitely improved but from time to time asterisk still sends updates and reinvites with new rpid information.. |
| 07:57.50 | bchamberlain | running 11.7 |
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| 08:11.38 | Geek-Linux | Deal All, Hope you all are fine. I am facing an issue. asterisk send invites, from send the respond with 100, and in the mean while it send reinvites. how to increase the reponse wait time in asterisk , so that it only acknowledge the reponse instead of sending reinvites. |
| 08:14.22 | WIMPy | bchamberlain: sendrpid=no |
| 08:15.06 | bchamberlain | hey WIMPy - thanks - I do need caller ID being sent just no updates.. T |
| 08:16.39 | WIMPy | You still have the from:. |
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| 08:17.28 | bchamberlain | WIMPy - true - but the carriers we deal with want rpid or pai only. |
| 08:17.54 | Geek-Linux | Deal All, Hope you all are fine. I am facing an issue. asterisk send invite message in SIP ITSP , telco respond with 100, and in the mean while it send reinvites. how to increase the reponse wait time in asterisk sip invitation , so that it only acknowledge the response instead of sending reinvites |
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| 08:21.15 | bchamberlain | WIMPy they don't use the from for asserting caller ID. Is there nothing else that can be done in v11 to stop connected line updates from being sent to trunks? |
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| 08:32.21 | Geek-Linux | Hope you all are fine. I am facing an issue. asterisk send invite message in SIP ITSP , telco respond with 100, and in the mean while it send reinvites. how to increase the reponse wait time in asterisk sip invitation , so that it only acknowledge the response instead of sending reinvites |
| 08:33.50 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:40.32 | chees0 | Geek-Linux: I'm just guessing here, but type in asterisk "sip show peer xxx", where xxx is your sip user and check out the sess-expires -value. more info about it: http://doxygen.asterisk.org/trunk/sip_session_timers.html |
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| 08:51.57 | chees0 | Geek-Linux: hmm... i think i jumped to conclusions too quickly. are you referring to https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance (third picture on page) the time between f6 and the next invite? |
| 08:54.21 | Geek-Linux | chess0: Thanks for your response. let me check.. i will update you in a while |
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| 08:58.53 | Geek-Linux | chees0: Now are you proposing me to set directmedia to outgoing ? in my sip.conf globals |
| 09:00.37 | Geek-Linux | but in my case i dont have multiple asterisk servers included. i only have an asterisk server which is connected to telco VOIP switch with a SIP ITSP. |
| 09:03.30 | chees0 | if you must have directmedia, then try different values for it. |
| 09:05.06 | chees0 | If i understood your problem correctly, there has been discussion about adding a delay value to the reinvite for directmedia. Unfortunately it didn't go well: http://lists.digium.com/pipermail/asterisk-dev/2012-August/056509.html and http://markmail.org/message/hzalysedlhd5v7s4 |
| 09:05.53 | chees0 | scratch that last directmedia, i meant "delay value to the reinvite" |
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| 09:17.09 | chees0 | Geek-Linux: let us know if you get it running |
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| 09:24.24 | skrusty | morning all |
| 09:24.32 | skrusty | chees0: you get it up and running on 11 ok? :) |
| 09:25.27 | chees0 | skrusty: didn't try it yet. got other projects atm |
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| 09:31.04 | chees0 | it's reverted and apt-get is fine. now just to get opus working :P |
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| 09:37.59 | skrusty | :) |
| 09:44.41 | mirela666 | http://www.dodaj.rs/f/M/St/1QMEvLZR/anybepb460sa.gif |
| 09:44.50 | mirela666 | Telephony was powerfull back in days |
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| 09:48.50 | skrusty | lol |
| 09:49.52 | skrusty | im sure people don't listen to the announcement at the start of a call.. otherwise i wouldn't keep getting calls for screwfix! :@ |
| 09:50.07 | chees0 | gif reminds me of http://bash.org/?4281 |
| 09:53.58 | chees0 | what announcement? |
| 09:54.51 | skrusty | the annoucenment at the start of calls to my business number! :) |
| 09:55.45 | chees0 | i haven't called you for a while, remind me a little bit what it consists of? |
| 09:55.57 | chees0 | *could you remind* |
| 09:56.02 | chees0 | *would* |
| 09:57.13 | skrusty | it doesn't say "welcome to screwfix"! :D |
| 10:00.48 | chees0 | does it say "if you're trying to reach screwfix, this is the wrong number and please do not hold"? |
| 10:01.28 | skrusty | no, but that's implied ;) |
| 10:03.17 | chees0 | i remember a buddy of mine had his waiting tone set to this old melancholic song about horses and lumberjacking. so like 8 years ago i locked myself out and i was trying to get my spare keys from him. To this day i know the lyrics by heart |
| 10:05.56 | skrusty | hehe :) |
| 10:16.42 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 10:17.02 | Blashyrkh | i dont quite get what the , n+1 for example in prorites does |
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| 10:20.49 | chees0 | what url? |
| 10:21.53 | Blashyrkh | what url? |
| 10:24.30 | chees0 | i mean can you show us what example you are referring to? |
| 10:24.44 | Blashyrkh | i just read here http://www.voip-info.org/wiki/view/Asterisk+priorities |
| 10:24.49 | chees0 | that url |
| 10:25.02 | Blashyrkh | but i dont quite understand what the arbitrary increment does |
| 10:26.42 | chees0 | others correct me if i'm wrong, but the n is just a number that gets incremented after each rule line |
| 10:27.12 | chees0 | if you have file that has 5 lines, the first is 1 and the others are n |
| 10:27.29 | chees0 | the computer gets the sequence as 1,2,3,4,5 |
| 10:27.57 | chees0 | if however the fourth line had n+2, it would show up to the computer as 1,2,3,6,5 |
| 10:28.30 | chees0 | right guys? |
| 10:29.17 | WIMPy | Yes, But I can't think of any use for that. |
| 10:31.05 | WIMPy | Well, not easily ;-) |
| 10:32.11 | chees0 | i see it as an stepping stone after the introduction of "n" and before inventing labels |
| 10:33.25 | kaldemar | since priority jumping has been gone for ages now, that's really just making things harder to understand. |
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| 11:38.49 | makmak78 | Hi! just posted a question in the asterisk-dev channel. it probably is better to post it in here. im getting "undefined symbol: __sync_bool_compare_and_swap_4" when trying to load app_konference 2.5 using asterisk 1.4.44 |
| 11:39.02 | makmak78 | any thoughts on why im getting that error |
| 11:41.26 | WIMPy | You didn't buid it for the Asterisk version you're running. |
| 11:42.40 | WIMPy | "version" might be misleading. It needs to be the exact configuration. |
| 11:42.45 | makmak78 | im am pretty sure i did |
| 11:42.53 | makmak78 | i pointed to the correct location in the makefile |
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| 11:44.55 | WIMPy | Was it a fresh install or did you upgrade? |
| 11:47.28 | makmak78 | fresh install |
| 11:49.38 | WIMPy | Might be easier to upgrade to a current Asterisk with ConfBridge than waiting for someone who remembers konference. |
| 11:50.08 | makmak78 | yes perhaps. thanks anyway |
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| 12:48.02 | makmak78 | i had to set march=i686 |
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| 13:03.23 | nakanik | Hi everyone. I want to integrate asterisk with our messaging tool. Have none asterisk knowledge. Are there any kind of web hooks/events sto that I can have asterisk call a URL upon starting a call and upon hanging a call (along with the operator's name)? Any general direction I should look into? |
| 13:04.43 | mirela666 | nakanik: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial has option URL |
| 13:04.57 | mirela666 | and after dial you can call url for disconnect |
| 13:05.41 | nakanik | thanks mirela666. Is this "cmd dial" an integral part of Asterisk or an add-on I should ask to be installed to my sysadmin? |
| 13:05.57 | mirela666 | Dial is asterisk application |
| 13:06.15 | mirela666 | basic one for making call |
| 13:06.32 | nakanik | oh so most probably we already use it to make calls |
| 13:06.43 | mirela666 | and when dial is finished (call hanged up) you can call SendURL |
| 13:06.54 | mirela666 | 100% |
| 13:07.01 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 13:07.25 | mirela666 | or you can make your external app and catch AMI events |
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| 13:07.44 | nakanik | perfect. and can I pass to the urls local parameters such as an operator's name (or how is called in asterisk terminology)? |
| 13:07.53 | mirela666 | http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 13:08.04 | mirela666 | yes |
| 13:08.25 | nakanik | Regarding AMI - If I got it correct, I have my application running and in constant connection with the AMI and I get notified upon specifc events? |
| 13:08.37 | mirela666 | yes |
| 13:08.42 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:08.51 | mirela666 | or you iniciate action and read the response |
| 13:09.01 | nakanik | so the 1st approach is less obtrusive for simpler needs like mine (I guess?) |
| 13:09.11 | mirela666 | yes i guess |
| 13:09.31 | nakanik | thanks mirela666 - you gave me excatly what I needed in 5 minutes :) |
| 13:09.54 | mirela666 | np :) |
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| 14:12.41 | lroe | Is there another company other than digium that makes asterisk appliances? (specifically larger than switchvox) |
| 14:15.18 | [TK]D-Fender | plenty |
| 14:15.57 | [TK]D-Fender | Fonality, ScopServ, Xorcom has a number of solutions |
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| 14:29.21 | Bkhan_ | Hi to all |
| 14:29.50 | Bkhan_ | My issue was when call come from trunck DTMF for voice mail not working |
| 14:30.11 | Bkhan_ | DTMF on ivrs are working fine |
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| 14:35.33 | *** join/#asterisk carrar (~tim@osburn.jp) |
| 14:36.41 | [TK]D-Fender | DTMF isn't any different for one vs another |
| 14:36.49 | Katty | good morning fender bender. |
| 14:36.50 | [TK]D-Fender | It is not part of the application, it's part of the channel |
| 14:36.51 | Katty | hugs carrar |
| 14:38.20 | [TK]D-Fender | Katty: Mew. |
| 14:39.12 | Katty | [TK]D-Fender: how're you dear? |
| 14:40.37 | carrar | Hi KattyRoooo |
| 14:40.44 | carrar | hugs Katthy back |
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| 14:43.23 | [TK]D-Fender | Katty: *barg* headache.. need drugs and it's still morning... |
| 14:43.49 | Katty | [TK]D-Fender: stress headache? |
| 14:45.05 | Bkhan_ | D-Fender> what you say in this scenario? |
| 14:45.44 | [TK]D-Fender | Bkhan_: The same thing anyone would... go look at those 2 calls very closely |
| 14:46.02 | Bkhan_ | Is it carrier issue |
| 14:46.33 | Bkhan_ | ok thanks |
| 14:46.42 | [TK]D-Fender | Bkhan_: We have seen nothing, don't expect us to know whats wrong |
| 14:46.52 | [TK]D-Fender | [09:36][TK]D-FenderDTMF isn't any different for one vs another |
| 14:46.59 | Bkhan_ | let me monitor both calls |
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| 14:48.03 | Bkhan_ | one thing i can do use SIPDtmfMode in voicemail context |
| 14:48.35 | Bkhan_ | and check many modes |
| 14:49.07 | [TK]D-Fender | no. |
| 14:49.15 | [TK]D-Fender | You don't seem to be comprehending. |
| 14:49.34 | [TK]D-Fender | IVR is Voicemail use the CHANNEL's DTMF by whatever it was configured to use |
| 14:49.45 | [TK]D-Fender | this is not something to change in the middle of the dialplan |
| 14:50.01 | [TK]D-Fender | s/is/and |
| 14:56.55 | Katty | fender, dear. |
| 14:57.00 | Katty | be nice to the children. |
| 14:58.30 | Chainsaw | Fender's speech patterns remind me of Mr Torgue. |
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| 15:32.06 | boom^time | Does it seem plausible that if you accidentally left an asterisk connection up having it dump info on 12 calls in high verbose color that it would cause the CPU load to spike from like a .3 average up to 4.0+? |
| 15:37.15 | nakanik | If all I care is a person's status, i.e. on call or free, which events (AMI) I am interested in? I want to create a new AMI account on our production server which is safe, i.e. has only read access to the events I care about. Is it just Agent events (AgentConnect and AgentComplete)? |
| 15:38.18 | pabelanger | nakanik, loaded question, depends what you are building. |
| 15:38.42 | [TK]D-Fender | Also depends on when you care |
| 15:39.03 | nakanik | I want to use something like adhearsion to get notified ASAP when one of my support representatives get busy and when they get free |
| 15:39.16 | nakanik | or any other AMI library in Ruby |
| 15:39.34 | pabelanger | so, you want to monitor your queue members |
| 15:39.36 | pabelanger | in app_queue |
| 15:39.44 | pabelanger | but how are you adding your members? |
| 15:39.58 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 15:40.04 | nakanik | no clue. i thought this is already configured on asterisk (?) |
| 15:40.10 | pabelanger | no |
| 15:40.43 | pabelanger | So, what you are asking is difficult to reply to. You could need QueueMemberStatus, PresenceStatus or something custom made |
| 15:41.42 | nakanik | ok i need to read some more it seems :) |
| 15:41.43 | nakanik | thanks |
| 15:41.45 | pabelanger | FWIW: we use all the events in app_queue and coddled together a dashboard system for site supervisors: https://twitter.com/pabelanger/status/398507630668177409/photo/1 |
| 15:42.19 | pabelanger | that was all HTML5 with web sockets |
| 15:44.02 | [TK]D-Fender | pabelanger: I push my queue and extension stats right to my CSR's phones' screens |
| 15:44.20 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 15:45.39 | pabelanger | Ya, different ways to skin the cat. The dashboard also allows a manager CRUD operations for app_queue and queue members. Plus statistics |
| 15:54.36 | skrusty | pabelanger: nice and clean :) |
| 15:55.28 | pabelanger | skrusty, yar, it's just the bootstrap css from twitter |
| 15:55.38 | skrusty | yeah, i love bootstrap :) |
| 15:55.46 | skrusty | saves me having to do almost any design :) |
| 15:55.55 | skrusty | and being a programmer, that's a big plus :D |
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| 15:59.18 | pabelanger | skrusty, Ya, I am terrible at html and CSS. Have people on the team that do it, however I'm a fan of minimal |
| 15:59.26 | pabelanger | seems to work well |
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| 16:36.56 | cusco | hey there... |
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| 17:02.51 | cusco | so I'm testing a cisco SPA303 |
| 17:03.05 | cusco | and it is on a different network ... |
| 17:03.53 | cusco | in the configuration webpage I activated the option nat mapping... |
| 17:04.05 | cusco | ok and I found advanced options to change nat parameters.. let me try |
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| 17:12.25 | Penguin | cusco: You still haven't learned the rules of using NAT and asterisk? |
| 17:12.35 | Penguin | After all this time, I thought you'd pick that up. |
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| 17:29.18 | peterthesing | hi i am using asterisk 12 on opensuse 13.1 and i get no visible traffic on udp port 5066. what am i doing wrong? |
| 17:29.35 | Penguin | Looking at the wrong port. |
| 17:29.49 | Penguin | SIP = UDP 5060 |
| 17:30.25 | peterthesing | what port should i be looking for? |
| 17:31.02 | peterthesing | ok |
| 17:31.36 | WIMPy | Or more precise the port you have configured. |
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| 17:37.51 | peterthesing | the default port is UDP 5060 but still no traffic |
| 17:38.25 | peterthesing | i will put it in anyway |
| 17:38.53 | WIMPy | Is there any reason for traffic? |
| 17:39.09 | Penguin | I would need more information to know what your problem is. |
| 17:39.40 | [TK]D-Fender | We don't know of any reason you should see traffic yet. |
| 17:39.44 | WIMPy | What problem? |
| 17:41.53 | newtonr | peterthesing, Is a SIP client calling Asterisk? Do you have Asterisk calling outbound? How are you looking for traffic? , etc |
| 17:42.44 | peterthesing | i have done reload and restart but still no succes |
| 17:43.18 | Penguin | Several things will cause traffic. Doing none of those things will get you NO traffic. |
| 17:43.41 | Penguin | The mere presence of asterisk will not generate traffic. |
| 17:45.02 | peterthesing | i have no sip devices calling asterisk but in webmin i can check wether somethings are going on regarding asterisk |
| 17:45.19 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 17:45.39 | WIMPy | If you don't tell it to do anything, that's what it will do: Nothing. |
| 17:46.03 | WIMPy | Well, sit there waiting for others to do something. |
| 17:47.10 | newtonr | peterthesing, you don't have anything calling Asterisk, and you don't know that Asterisk is doing anything... what in particular are you looking for? |
| 17:47.13 | drmessano | Most apps will do nothing when they are not configured or interacted with, except Microsoft products |
| 17:48.45 | peterthesing | just to see if there is traffic or activity on port UDP 5060 |
| 17:49.09 | drmessano | There won't be |
| 17:49.15 | drmessano | You've not told it to do anything |
| 17:49.24 | drmessano | or anything to do anything with its anything |
| 17:49.40 | newtonr | peterthesing, Is it... part of a class project or something? Did you configure the Asterisk box from scratch, or did someone else? |
| 17:49.43 | [TK]D-Fender | peterthesing: Why would there be traffic if you didn't set Asterisk to talk to something, and you didn't set anything up to contact Asterisk? |
| 17:50.03 | peterthesing | if there is no activity how can i connect? |
| 17:50.17 | Penguin | Connect what to what? |
| 17:50.24 | drmessano | lol |
| 17:50.42 | [TK]D-Fender | Who's on first? |
| 17:51.01 | Penguin | I Don't Know. |
| 17:51.13 | drmessano | Will it go round in circles, will it fly high like a bird up in the sky |
| 17:51.56 | peterthesing | in the yast application i selected asterisk and it did the rest |
| 17:52.06 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:52.11 | Penguin | Great! So asterisk is INSTALLED! |
| 17:52.25 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 17:52.52 | peterthesing | but now how to configure it! |
| 17:52.58 | WIMPy | Do you also start Apache and expect it to flood the Internet with web pages without anyone requesting them? |
| 17:53.03 | WIMPy | ~book |
| 17:53.03 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:53.04 | drmessano | peterthesing, config files! |
| 17:53.12 | Penguin | Time to find a text editor that you know how to use. |
| 17:53.34 | WIMPy | And read that book befor changing them. |
| 17:53.56 | drmessano | Then, and only then, will you have stuff - Me |
| 17:54.08 | WIMPy | And getting a phone might help a lot. |
| 17:54.33 | newtonr | peterthesing, the Definitive Guide is a great place to start. Also reading on the wiki may help you.. https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality |
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| 17:56.16 | peterthesing | ok i will read that thanks for helping me out |
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| 19:15.20 | agoodm | Hi, where can I locate documentation relating to troubleshooting jitter buffers? |
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| 19:37.22 | newtonr | agoodm, I don't know that any documentation of that sort exists. Other than documentation for individual configuration options, applications, functions, etc and things users have written out on the net. |
| 19:37.50 | newtonr | Of course there may be something I'm not aware of! |
| 19:38.20 | newtonr | file, ^ ? |
| 19:38.42 | file | I got nothin' |
| 19:41.46 | pabelanger | well, if you need to use jitterbuffers, I would be looking at your network first |
| 19:42.04 | pabelanger | IMO: jitterbuffers are a last resort to a bad situation. |
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| 19:56.30 | agoodm | pabelanger, I have looked at my network. we have quality of service on the network however when quality of service is managing the connection there is about 50ms of latency jitter (20ms idle up to about 75ms under load) |
| 19:57.22 | agoodm | thus I hear chops in audio when transfers start, a 100ms fixed jitter buffer should prevent those chops, however I hear little to no improvement in the call with it enabled |
| 19:58.20 | agoodm | newtonr, ^^ |
| 19:59.12 | agoodm | without QoS calls are totally useless, there is 400+ms of jitter and you get too much packet loss |
| 19:59.24 | agoodm | the QoS manages to keep packet loss to 0 for voip packets |
| 19:59.26 | mjordan | how are you enabling the jitter buffer? |
| 19:59.32 | mjordan | and what version of Asterisk are you on? |
| 19:59.37 | agoodm | 1.8LTS |
| 19:59.41 | mjordan | hm |
| 20:00.05 | agoodm | in sip.conf general section: jbenable = yes jbimpl = fixed jbmaxsize = 100 |
| 20:00.12 | agoodm | on separate lines of course |
| 20:00.15 | mjordan | So. Jitter buffers enabled in the channel drivers are on the side of the channel that you wouldn't expect - they're on the write side, as opposed to the read side |
| 20:00.45 | mjordan | there's a good chance you've put the jitter buffer on the channel/peer that has the jitter - which, paradoxically, isn't where you want it. You want it on whoever they are talking to. |
| 20:00.47 | agoodm | as in the jitter buffer must be on the 'receiving' side? |
| 20:01.04 | mjordan | We fixed this in 10+ via func_jitterbuffer, which puts the jitter buffer where it should have been the entire time. |
| 20:01.13 | agoodm | ok well thats good, since I also run the voip provider I have enabled the same settings on the sip trunk server :) |
| 20:01.16 | pabelanger | Ya, was going to say that |
| 20:01.25 | pabelanger | you likely had jb on the wrong side of the flow |
| 20:01.35 | mjordan | I'd recommend, if it's an option, taking a look at Asterisk 11. The JITTERBUFFER function is handy |
| 20:01.39 | agoodm | so I control both sides... |
| 20:02.12 | mjordan | not to mention with pre-dial in Asterisk 11, you can definitely always put the jitter buffer exactly where you want it/need it |
| 20:02.24 | agoodm | upgrading the hardware is in the pipeline for this client, ive got asterisk 11 i think on the new box but the internet connection is currently faulty there so cant move asterisk yet |
| 20:04.37 | mjordan | agoodm: there's a bit of information on this on the wiki here: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5243109 |
| 20:05.07 | mjordan | and this notes the addition of the JITTERBUFFER function: https://wiki.asterisk.org/wiki/display/AST/New+in+10#Newin10-DialplanFunctions |
| 20:05.13 | mjordan | granted, both pages are a little terse |
| 20:05.20 | agoodm | ok on the receiving side I have got the following in sip.conf general... |
| 20:06.05 | mjordan | basically, in your version of Asterisk, you need to enable jitter buffers on whatever the jittery connection is talking to |
| 20:06.42 | agoodm | ok so in my test case I am talking to a asterisk server which sits behind qos at my house |
| 20:06.52 | agoodm | its known good working solution, no call quality issues |
| 20:07.18 | agoodm | it has had the jitter buffer enabled earlier on with the same settings as I used at the clients site |
| 20:07.30 | agoodm | how can I double check the jitter buffer is being used? |
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| 20:11.02 | agoodm | i mean the calls are perfectly usable, just every now and then youll hear kind of a bit of a slurrrr in the audio... is that really consistent with jitter? |
| 20:11.47 | BarthezZ | could be agoodm |
| 20:12.01 | BarthezZ | start tcpdump, import the dump to wireshark on your workstation, and do an RTP analyse :) |
| 20:14.10 | agoodm | will that show me that jitter buffer is active, or total jitter or? |
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| 20:16.10 | BarthezZ | agoodm it will give you numbers on the amount of jitter, packet loss, etc. you have per call :) |
| 20:16.24 | BarthezZ | let's say if you capture for an hour, you could get an average on your problems |
| 20:16.46 | agoodm | brill :) |
| 20:17.10 | BarthezZ | ? |
| 20:17.19 | agoodm | what ill do is tomorrow during peak system usage I will make a test call of an hour with tcpdump logging |
| 20:17.30 | agoodm | then ill rtpanalyse the file to see where we stand on average :) |
| 20:17.49 | BarthezZ | or, if you want that longterm... check out voipmonitor |
| 20:18.06 | BarthezZ | I'll install it for network-troubleshootings on some pbxes' |
| 20:18.17 | *** join/#asterisk VaGoNeTaS (~debianzot@186.67.82.18) |
| 20:18.20 | VaGoNeTaS | hey everyone |
| 20:18.34 | VaGoNeTaS | im looking for some help |
| 20:18.49 | VaGoNeTaS | im trying to configure an virtual fax on Eastix |
| 20:19.07 | VaGoNeTaS | i've tried on Elastix 2.2.0, 2.3.0 and 2.4.0 versions |
| 20:19.09 | pabelanger | T.30 over SIP? |
| 20:19.20 | VaGoNeTaS | in all versions, i got fax tone working |
| 20:19.34 | pabelanger | you likely want #elastix for elastix specific questions |
| 20:19.34 | VaGoNeTaS | but at the moment when i try to receive a fax from a real fax machine, it doesnt go through |
| 20:19.44 | VaGoNeTaS | it keeps all the time "connecting" |
| 20:19.49 | pabelanger | VaGoNeTaS, ^ |
| 20:19.51 | VaGoNeTaS | and yes, its over SIP |
| 20:20.18 | pabelanger | You should be using T.38 over SIP, T.30 is not going to work well for you |
| 20:20.34 | VaGoNeTaS | what u mean T.30 or T.38 dude? |
| 20:20.48 | pabelanger | ~book |
| 20:20.48 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:21.04 | pabelanger | VaGoNeTaS, ^ check out the section regarding faxing |
| 20:21.10 | pabelanger | they are fax protocols |
| 20:21.16 | VaGoNeTaS | i see |
| 20:21.27 | VaGoNeTaS | i may be using T.30... |
| 20:21.31 | VaGoNeTaS | might* |
| 20:21.39 | VaGoNeTaS | ill check that out |
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| 20:22.56 | pabelanger | Anybody ever used nuances mobile interface for TTS? |
| 20:23.08 | pabelanger | Specing out a project and looking at using it with asterisk |
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| 20:32.20 | rrittgarn | with mixmon recording (.wav files) from features.conf I'm not getting a duration on the files when i use soxi to inspect them. They play with VLC / Chrome, but no in WMP. Also because there is no duration, sox can't convert them correctly. Lame Can, but i don't get audio in WMP again. Anyone run into this / know what I'm missing? |
| 20:35.13 | *** part/#asterisk mjordan (~matt@nat/digium/x-rtpnzymgjhgifjkb) |
| 20:35.41 | rrittgarn | nevermind... figured out my problem... I'm trying to send them before they are 100% written |
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| 20:44.15 | *** join/#asterisk dingle (dingle@secretpolice.net) |
| 20:45.19 | dingle | hello everyone. could someone please recommend a desktop computer to run asteriskNOW on? i am thinking of a p4, 2gb ram, 40gb hdd, cd-rom. would this work well? |
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| 20:45.46 | pabelanger | dingle, yes |
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| 20:46.36 | dingle | ok, do i just need 1 ethernet port or 2? 1 for internet access and the other to goto the phone? |
| 20:46.48 | newtonr | dingle, just the one |
| 20:46.50 | dingle | or can i just connect the 1 ethernet port into my switch as well as the phone |
| 20:46.51 | dingle | ok thanks |
| 20:48.43 | dingle | now if i can just find someone to donate the computer to me. it seems like a machine a lot of people would just throw out, maybe i could place a wanted ad on craigslist |
| 20:50.30 | newtonr | LOL |
| 20:50.49 | dingle | the thing is only worth like $50 tops |
| 20:51.47 | dingle | i have the UC |
| 20:52.45 | Qwell | I had that once... worst rash ever. |
| 20:53.24 | dingle | cool |
| 20:53.40 | dingle | i am new to this, so i apologize if my questions annoy you |
| 20:53.44 | dingle | i am learning |
| 20:54.14 | pabelanger | dingle, make sure you use the book |
| 20:54.17 | pabelanger | ~book |
| 20:54.17 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:54.39 | dingle | yeah, i have the link to it |
| 20:54.41 | dingle | thank you |
| 20:55.45 | dingle | is vitelity a good provider to go with for the voip? its only $8 a month unlimited |
| 20:59.35 | dingle | i have this phone: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps8537/product_data_sheet0900aecd8069bd58.html |
| 20:59.48 | dingle | do they make wireless phones so i can put one in my bedroom as well? |
| 21:02.56 | rrittgarn | yealink makes a portable unit that isn't terrible and is reasonably priced... compared to some of the other units |
| 21:05.51 | dingle | so its a cordless phone not wireless? |
| 21:06.39 | [TK]D-Fender | cordless = wireless, cord = wire |
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| 21:07.48 | dingle | uh |
| 21:07.50 | dingle | right |
| 21:08.39 | [TK]D-Fender | Perhaps a rephrasing is in order |
| 21:08.49 | *** join/#asterisk edwin_quijada (~edwin_qui@186.1.86.190) |
| 21:08.51 | edwin_quijada | Hi! |
| 21:09.15 | edwin_quijada | What app can I use and format to record prompts for an IVR ? |
| 21:09.48 | edwin_quijada | I am trying to use Audacity but I dont know what format asterisk knows to play |
| 21:10.28 | dingle | i'm no expert, but i bet it will play a .wav file |
| 21:11.10 | edwin_quijada | wav doesnt work it was the first that I tested |
| 21:11.17 | wdoekes | 16bit 8khz mono |
| 21:11.33 | [TK]D-Fender | wav is a generic container that could hold any number of bitrates, depths, channels, etc |
| 21:11.43 | edwin_quijada | GSM 6.1 WAV Audio MObile |
| 21:13.40 | edwin_quijada | I am trying to use this wuth 8khz and mono aural but asterisk tell me format_wav.c:94 check_header_fmt: Not a wav file 49 |
| 21:13.40 | edwin_quijada | [Jan 30 09:26:57] WARNING[4719]: file.c:391 fn_wrapper: Unable to open format wav |
| 21:14.58 | [TK]D-Fender | WAV != wav |
| 21:16.51 | edwin_quijada | Ohh |
| 21:16.57 | edwin_quijada | let me check again then |
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| 22:00.06 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:01.53 | b3gg1n3r | Hi everyone, I am a beginner in asterisk. I want to forward long distance pstn calls, from one server to another server, through sip. Is there a good guide about it (for beginners). Please help. Thank you |
| 22:02.16 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 22:02.17 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 22:04.07 | pabelanger | b3gg1n3r, you'll want to read ~book |
| 22:06.12 | b3gg1n3r | I have gone through the officla documentation on asteriskdocs. Even though it shows how to connect PBXs through dundi for calls using extensions |
| 22:06.53 | b3gg1n3r | what i want is something like.. dial a pstn number through a remote asterisk pbx, so that I can avoid long distance charges of pstn |
| 22:07.38 | [TK]D-Fender | Make a pper to call the other box. Use it. |
| 22:07.51 | [TK]D-Fender | Have dialplan on that remote peer to process the call and go out |
| 22:08.58 | b3gg1n3r | what is pper |
| 22:09.10 | b3gg1n3r | sorry, just like my name shows, i am a beginner !! |
| 22:10.08 | [TK]D-Fender | peer* |
| 22:10.32 | b3gg1n3r | hmm ok thank you tk, let me give it a shot with the help of google !! thanks alot |
| 22:10.44 | [TK]D-Fender | DUNDi is something that can also be largely disregarded most of the time. |
| 22:11.36 | b3gg1n3r | dundi is already configured on my server and calls between extensions through dundi is working |
| 22:12.54 | b3gg1n3r | thank you very much tK |
| 22:12.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.174) |
| 22:18.21 | edwin_quijada | wich TTS can I use for asterisk in spanish with linux |
| 22:18.32 | edwin_quijada | No cepstral please |
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| 22:50.56 | bchamberlain | Hej - is it possible at all to stop asterisk 11.x from sending reinvites with caller ID information to trunks? I've tried directmedia=no canreinvite=no and rpid_update=no to no avail. |
| 23:00.07 | *** join/#asterisk Geek-Linux (~Geek-Linu@182.188.195.166) |
| 23:12.37 | Geek-Linux | Dear All, Hope you are fine.. i am facing and issue. When I use Asterisk Management Interface (Port 5038), and use Originate, it initialize 2 call at a single time with single command am i doing some thing wrong. detail are on the link http://pastebin.com/NrHzFYSj |
| 23:13.06 | Geek-Linux | but from the normal CLI there is no issue |
| 23:18.45 | Geek-Linux | any one.. ? |
| 23:26.38 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 23:27.26 | [TK]D-Fender | Geek-Linux: What is wrong with it? |
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| 23:39.14 | *** part/#asterisk bchamberlain (~brian98@unaffiliated/brian98) |
| 23:40.28 | Geek-Linux | [TK]D-Fender: when i use orginate command from CLI works fine, but when i use Manager Interface Two calls are intiated on the same number and then busy prompt.. dont know why |
| 23:41.43 | [TK]D-Fender | Channel: Local/XXXXXXXXXX |
| 23:41.48 | [TK]D-Fender | Exten: XXXXXXXXXX |
| 23:42.05 | [TK]D-Fender | You seem to be giving it the same number (assumed since you're hiding the EVIDENCE) |
| 23:42.46 | [TK]D-Fender | And that AMI call looks... really pointless |
| 23:43.22 | [TK]D-Fender | Doesn't look like any attention was paid as to where it was being pointed.... |
| 23:43.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.174) |
| 23:44.32 | [TK]D-Fender | Call john via dialplan. ANSWER IMMEDIATELY .. then actually dial out a peer to John. And while that is going on... do the EXACT same thing. So when john answers the first, the second gets back "busy" |
| 23:48.44 | [TK]D-Fender | Or more likely the first is actually "ringing" when the 2nd attempt is ack'd and then gets turned down because the telco can't pass an additional call while "ringing" |
| 23:48.56 | [TK]D-Fender | Which is what cells & ananolg would do. |
| 23:49.01 | [TK]D-Fender | analog* |
| 23:52.24 | Geek-Linux | [TK]D-Fender: Thank You for your response, problem is resolved. From extension i removed the Dial and every thing is OK, Now in channel i mentioned the number and ITSP name. Every thing works like a charm :D |
| 23:53.22 | [TK]D-Fender | Always check your targets :) |
| 23:53.38 | [TK]D-Fender | That "answer" in front is typically a BAD thing |
| 23:54.29 | [TK]D-Fender | You forced immediate processing before a "person" answered.... there are some reasons to do this... but the is processing the Exten side... before you actually called out |
| 23:56.22 | [TK]D-Fender | checkout time, BBL-ish (maybe) |
| 23:57.48 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
