| 00:01.26 | asterisk-user | everybody sleeping ? |
| 00:01.31 | asterisk-user | really ? nobody on ? |
| 00:01.39 | asterisk-user | OMG :( |
| 00:01.50 | almostworking | must be new to IRC, takes hrs sometimes |
| 00:02.03 | asterisk-user | yes something like that :P |
| 00:02.06 | almostworking | differnt time zone , differnt people..... etc |
| 00:02.15 | asterisk-user | yes yes :) my bad |
| 00:02.37 | almostworking | also , normal to ask before PM someone, just fyi , as some get quite upset, irrate about it |
| 00:02.48 | asterisk-user | oh my bad |
| 00:03.00 | asterisk-user | I'am sorry for that |
| 00:03.02 | almostworking | i dont know asterisk , its just one part of one of the things i need to get done . |
| 00:03.14 | asterisk-user | can i talk with you PM ? |
| 00:03.18 | asterisk-user | if you don't mind |
| 00:03.18 | asterisk-user | ? |
| 00:03.40 | almostworking | well, i guess, however might be helpful in here as sometimes people will join the conversation |
| 00:04.42 | almostworking | unless its somehow personal info, but not sure how that could be.... |
| 00:05.22 | asterisk-user | no ... i just to want to ask dumb questions and all the people leaf at me :P |
| 00:05.46 | asterisk-user | i need to integrate twilio with my asterisk IVR ... and i don't know how... |
| 00:05.55 | cryptic | asterisk-user: you're probably new to IRC. no one in here is paid to provide assistance, so ask in the channel. you'll get responses faster that way. alternatively, go to odesk.com or elance.com to hire someone. |
| 00:05.56 | asterisk-user | i will pay 200$ for the one that will help me |
| 00:07.05 | asterisk-user | so... can i get a little help please? |
| 00:07.21 | almostworking | you need in person help, $200 u can hire someone to come to your house and conslut, fix it etc . |
| 00:07.27 | almostworking | remote help is difficult. |
| 00:07.59 | almostworking | ive never used asterisk, is just part of programs i have tested like, FreeBPX |
| 00:08.29 | asterisk-user | is not difficult with teamviewer |
| 00:08.34 | almostworking | or PBX in a flash , other ones cant remember . |
| 00:08.56 | almostworking | i dont have team viewer , on my laptop runs #! |
| 00:09.02 | asterisk-user | anyway.. i don't know why the XML from twilio is not working .... |
| 00:09.46 | asterisk-user | i was wondering if somebody know how to do that |
| 00:10.33 | asterisk-user | for somebody who knows.... will take like 3 minutes .... |
| 00:14.23 | almostworking | u might try a channel on freenode that has something to do with the issue , like web design, or XML issues, .... these things not in my tool box. |
| 00:16.06 | almostworking | just a heads up , a general rule on IRC is not giving out personal info, that would include servers and credentials. if its a mission critical or limited time all that, asterisk-user , i would hire someone local , lots of people that consult for just VOIP stuff |
| 00:18.23 | asterisk-user | ok thanks |
| 00:18.45 | asterisk-user | I'am dumb but not 100 |
| 00:18.47 | asterisk-user | 100% |
| 00:19.55 | ChannelZ | Do they not speak SIP? |
| 00:20.15 | ChannelZ | I've not heard of Twilio and am unclear what it even is from their site |
| 00:26.44 | ChannelZ | shrugs and heads for home |
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| 00:29.03 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:33.47 | Iamnacho | malcolmd: need something? You Pinged me? |
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| 00:46.42 | oli-werk | anyone know of some irc channels more broadly related to voip? |
| 00:50.03 | ganbold_ | asterisk-user: maybe you need to provide some background info on what you are trying to do and exact problems you have |
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| 00:52.49 | ganbold_ | I did some twillio sms integration to some Rails site, but that is not really related to asterisk |
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| 00:53.50 | oli-werk | they have an api with documentation available |
| 00:54.01 | oli-werk | use an agi script or gosub() curl function |
| 00:55.08 | oli-werk | in some extension |
| 00:55.13 | oli-werk | and have your ivr point to that extension |
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| 02:57.55 | pjensen00 | hey, is someone there at this hour? |
| 02:58.17 | pjensen00 | I'm doing a LOCAL originate, but I"m getting double the channels spawned. |
| 02:58.55 | pjensen00 | As in Local/100@entry_point-0000002b;2 twice |
| 02:59.02 | pjensen00 | and the ;1 twice |
| 02:59.14 | pjensen00 | I'm not entirely certain what the heck is going on. |
| 03:03.43 | pjensen00 | this is the debug from the console http://pastebin.com/HZmsk8cm |
| 03:04.23 | pjensen00 | It doesn't look like I'm dialing twice. I see both of the local dials doing what they're supposed to be doing. |
| 03:05.50 | pjensen00 | Oh. nevermind. I'm seeing something bad happening. I just don't know what it is yet. :( |
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| 04:11.18 | pjensen00 | Aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaand nevermind. I figured it all out. The 'h' extension. Of course. OF COURSE. This is how I know it's late. |
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| 07:26.23 | xenol | Hi, I have multiple public IP addresses on my host and I need to set outgoing address for asterisk. Is that possible? |
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| 07:36.33 | ChannelZ | chan_sip can only bind to one or to all interfaces (in which case I don't actually know how it chooses an IP) |
| 07:38.12 | ChannelZ | pjsip can bind to multiple interfaces, and individual endpoints can be bound to whichever |
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| 08:11.40 | zapotah | is there a way for the ivr to require the caller to input a randomly generated, say three digit number, before continuing the call |
| 08:12.13 | zapotah | so that it generates a new required number for each call |
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| 08:23.50 | ChannelZ | sure |
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| 09:25.04 | zapotah | ChannelZ: that is highly unhelpful, i tried searching a bit but couldnt find an example for IVR scripting in this fashion. |
| 09:25.34 | zapotah | this is mainly to repel these automated phishing calls that seem to be on the rise |
| 09:26.31 | zapotah | a human can easily input three digits, a bot cannot |
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| 09:50.16 | _abc_ | Hello. I am looking for 1.6.22 and the only place I could find it is at http://olex.openlogic.com/packages/asterisk/1.6.2 . Is there perhaps a repo of older packages on digium.com? ftp links to ftp://digium.com/pub/... are broken, server is not running. I can understand that, but are the packages gone gone? |
| 09:54.09 | _abc_ | Err okay I just read the EOL notice... |
| 10:01.34 | _abc_ | daft |
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| 10:54.44 | din3sh | hey ppl |
| 10:54.57 | din3sh | anybody using grandstream phones in here? |
| 10:59.07 | kleszcz | yes |
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| 11:11.33 | din3sh | whats your general review of gxp phones? |
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| 13:38.34 | Prinni | Hello |
| 13:38.48 | Prinni | loves asterisk |
| 13:40.00 | Prinni | i have some security questions if anyone is interested in hearing them |
| 13:40.08 | Prinni | don't all shout at once |
| 13:40.41 | Prinni | watches the tumbleweed circle by... |
| 13:45.45 | Prinni | ok |
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| 13:46.44 | Prinni | I have two options for secure telephony (1) Use STU and STE encrypted desk handsets with the crypto-card modules, or (2) use Asterisk implemented with TLS and (Z)SRTP end-to-end encryption |
| 13:52.03 | malcolmd | asterisk doesn't do zrtp. it does do srtp. with srtp, you'll get encryption from the endpoint to asterisk, but not end-to-end between endpoints. the srtp keys will be contained in the tls signaling, and asterisk will have the tls signaling because it acts as a b2bua. |
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| 14:00.05 | dlynes | Does Digium no longer have a commercial asterisk server offering? I only seem to see switchvox promoted on their website |
| 14:03.02 | [TK]D-Fender | .... that is very precisely a "commercial asterisk offering" |
| 14:03.15 | [TK]D-Fender | They don't need 2 |
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| 14:11.40 | dlynes | Thought they had another one. I remember a couple of years ago there was a commercially supported version of asterisk that was several versions behind the open source version. |
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| 14:14.06 | dlynes | I guess that's a discontinued product, now? I think it came out before Digium purchased Switchvox |
| 14:14.59 | [TK]D-Fender | That was just solitary software w/o a GUI |
| 14:15.23 | [TK]D-Fender | what was and no hardware |
| 14:15.32 | dlynes | ah...so it didn't come with hardware |
| 14:15.33 | [TK]D-Fender | so basically a locked down asterisk only |
| 14:16.05 | Prinni | malcolmd: thats very helpful. thank you |
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| 14:17.33 | Prinni | so with asterisk configured with TLS and SRTP, you will get encryption between the sender phone and asterisk, then asterisk to the VoIP provider, then the VoIP provider to the receiver PABX, and receiver handset? |
| 14:17.52 | Prinni | obviosly would need to make sure the receiving end is configured to use TLS and SRTP |
| 14:18.22 | dlynes | [TK]D-Fender: thanks. Just started working for a digium reseller, and was looking to see if there were additional hardware options. I surmise I'll have to put in some feature requests for Switchvox 6 to get the features added that I'd like to see |
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| 14:24.53 | [TK]D-Fender | dlynes: No idea what their process is for feedback / custom... |
| 14:25.23 | dlynes | [TK]D-Fender: probably have to submit it through our channel manager |
| 14:25.37 | [TK]D-Fender | Go investigate... |
| 14:26.00 | dlynes | anyways..gotta run. Teaching programming to some kids in 1/2 hr |
| 14:46.35 | malcolmd | Prinni: right. with tls + srtp, it's hop to hop. if your voip provider hops onto the piston though, there's no encryption. |
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| 14:47.22 | malcolmd | hah.. pstn, not piston. thanks auto-correct |
| 14:52.18 | Prinni | oh lol |
| 14:52.21 | Prinni | but then would it fail |
| 14:52.31 | Prinni | would the servers inbetween be aware? |
| 14:52.35 | Prinni | and terminate te call? |
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| 17:18.29 | sung | any of you guys have 8.5(2) SIP firmware for Cisco 7970G ? |
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| 18:21.18 | bermanmk | is this the place to come with asterisknow install problems? |
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| 18:24.42 | bermanmk | I ran the asgterisknow install (from a usb drive if it matters) and it seems to have failed to actually install asterisk. |
| 18:24.51 | bermanmk | the install.log has a series of errors http://pastebin.com/q6V8yJMr |
| 18:25.06 | bermanmk | Iâve reinstalled and the same weird configuration results |
| 18:26.50 | malcolmd | there's a usb disk image installer here - http://downloads.freepbxdistro.org/AsteriskNOW/ (the .img files, not the .iso files) did you try it? |
| 18:27.20 | bermanmk | no, I just used unetbootin on the iso |
| 18:27.56 | bermanmk | Iâll try that instead |
| 18:28.00 | bermanmk | thanks |
| 18:28.16 | malcolmd | np. see if that clears it up. |
| 18:29.02 | Penguin | Are you installing from optical media to a hard drive? |
| 18:30.43 | *** join/#asterisk simonpatapon (~sime@modemcable078.201-37-24.static.videotron.ca) |
| 18:31.03 | bermanmk | no, a usb stick. |
| 18:31.44 | Penguin | You're going to have to be a little more clear with your answer. |
| 18:32.19 | bermanmk | installing to a hard drive from a 32gb usb drive. a sandisk one if it matters |
| 18:32.31 | lorsungcu_ | malcolmd: that customer with misbehaving D50s is having trouble right now - what logs can I provide that would help? |
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| 18:35.43 | malcolmd | PTSR from phone, packet capture from asterisk (though the second's probably not something you've got running all of the time). and, some knowledge of what they were doing before they noticed the issue and when they noticed the issue. and, if you're willing to manually provision a phone, outside of dpma, i've got something you can try on them. though if they require access to dpma-specific apps, like the parking app, then we're not yet r |
| 18:37.09 | lorsungcu_ | I should be able to get PTSR |
| 18:37.18 | lorsungcu_ | Might have a packet capture |
| 18:37.30 | lorsungcu_ | they were doing nothing; just noticed it wasn't working all of a sudden |
| 18:37.41 | Penguin | notices that malcolmd needs a better IRC client -- one that spans multiple posts rather than truncating a single one. |
| 18:37.47 | lorsungcu_ | Pressing the speaker button lights it red, but no dialtone, etc. |
| 18:39.00 | malcolmd | yeah, similar to before. post that to the support ticket response along with anything else you can collect. |
| 18:39.15 | malcolmd | Penguin: i haven't checked to see if there are any convos updates lately. |
| 18:39.55 | Penguin | That long one ended with "then we're not yet r" and you left us hanging. |
| 18:40.52 | malcolmd | nice |
| 18:41.32 | file | malcolmd, don't leave me hangin' bro |
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| 20:31.23 | bermanmk | ok, so it installed more appropriately from usb, but I have a new problem: apache fails to start at boot: |
| 20:31.37 | bermanmk | it warns âStarting httpd: Warning: DocumentRoot [/var/www/html/aastra/] does not existâ, |
| 20:31.38 | *** join/#asterisk PaybackTony (~Adium@2601:7:700:23:2db0:b8b6:1265:240) |
| 20:31.58 | bermanmk | then [Tue Mar 24 16:28:54 2015] [warn] _default_ VirtualHost overlap on port 80, the first has precedence |
| 20:32.14 | PaybackTony | Anyone here have success with Asterisk + WSS at any kind of scale? |
| 20:33.05 | *** join/#asterisk mikealeonetti (~mike@ool-2f141af4.dyn.optonline.net) |
| 20:35.04 | bermanmk | my /var/log/bootlog: http://pastebin.com/pV2h3vm9 |
| 20:35.06 | mikealeonetti | Why would I be getting "Retransmission timeout reached on transmission" errors when connecting my SIP softphone to asterisk via localhost? |
| 20:44.52 | *** join/#asterisk kb3ien (~kb3ien@ool-2f121857.dyn.optonline.net) |
| 20:46.07 | kb3ien | Having problems with linphone (android) not accepting video through asterisk. Linphone is asking for video RTP on port 0. Which asterisk won't. |
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| 20:46.11 | bermanmk | though of course thatâs probably the wrong log file to share |
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| 20:51.54 | kb3ien | m=video 0 RTP/AVP 0 |
| 20:53.31 | WIMPy | Looks like no video to me. But I don't know much SIP. |
| 20:55.16 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:55.18 | kb3ien | is there something that asterisk might be sending to prompt this ? |
| 20:55.45 | kb3ien | The invite said this : m=video 16462 RTP/AVP 99 |
| 20:59.16 | bermanmk | ahh, what is this schmoozecom business? |
| 21:00.11 | mjordan | kb3ien: a port of 0 indicates the UA is declining the media stream. |
| 21:00.32 | mjordan | that's the correct way for something to reject a media stream in an offer without rejecting the entire offer with a 488 |
| 21:00.50 | kb3ien | http://pastebin.com/6emaK3eQ it seems to be declining before looking at the media, I'll try other codecs. thanks. |
| 21:01.59 | kb3ien | nevermind the codec is in the invite. |
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| 21:23.12 | bermanmk | if I just do an ubuntu install and apt-get to install asterisk can I get it working without all this schmoozecom upsell garbage everywhere? |
| 21:24.44 | robmal | If you can read - yes. |
| 21:25.09 | [TK]D-Fender | bermanmk, Shmoozecom ... is FREEPBX stuff. That is not ASTERISK |
| 21:25.14 | WIMPy | bermanmk: Where did you find that? |
| 21:26.52 | bermanmk | apparently in the 5 years since I had to do any maintenance on my asterisk system the ecosystem got way more inbred. |
| 21:27.42 | *** part/#asterisk kharwell (kharwell@nat/digium/x-giskubscwtaubeta) |
| 21:28.27 | [TK]D-Fender | wonders what qualifies as "inbred" in this case |
| 21:29.20 | robmal | Recent google results show only complicated examples :-( |
| 21:29.50 | [TK]D-Fender | robmal, for? |
| 21:30.09 | robmal | For anything asterisk, since it's "inbred" now. |
| 21:30.50 | [TK]D-Fender | robmal, the usage of that word is still in question... |
| 21:31.25 | [TK]D-Fender | robmal, Also ... I'm not sure wwhy anyone should care about random Google results. So they haven't done a recent web crawl...... |
| 21:31.41 | [TK]D-Fender | "Google" is not the "asterisk community" |
| 21:34.03 | robmal | I think he ment that the 'asterisk howto' google query now leads to a quickstart guide in pdf, with screenshots, explaining everything. So it's inbred because it's boring. |
| 21:34.41 | [TK]D-Fender | inbred and boring are 2 complete different words. |
| 21:35.15 | robmal | Very observant. |
| 21:36.19 | robmal | OffTopic, since i have you online - can you show me your polycom provisioning config or at least tell me what features are used most frequently? |
| 21:36.44 | [TK]D-Fender | What kind of "features" are you referring to? |
| 21:37.40 | robmal | I've got BLF and contact directory figured out, trying to automate enabling screen capture. |
| 21:37.52 | robmal | What else is there? |
| 21:38.15 | *** join/#asterisk ryan42 (znc@stargate.rlntx.net) |
| 21:38.16 | [TK]D-Fender | What is "screen capture"? |
| 21:38.42 | ryan42 | hi all. I am having an intermittent problem answering inbound calls on my asterisk system |
| 21:38.45 | robmal | <up up.screenCapture.enabled="1" /> |
| 21:38.46 | [TK]D-Fender | On top of the rest I use the MicroBrowser, custom ring-tones, auto-answer for paging, etc |
| 21:39.10 | ryan42 | most of the time it works fine but about 25% of the time there's no audio and the call drops after 30 seconds or so. I have captured this happening here: http://pastebin.com/9PR9M2Vx |
| 21:39.23 | robmal | You can view the screen of the phone via http. Very useful, but needs enabling after each restart. |
| 21:39.55 | ryan42 | it seems like this is a source of issue: [2015-03-24 16:34:57] WARNING[2448]: chan_sip.c:4047 retrans_pkt: Retransmission timeout reached on transmission 5dfa1258-4d10-1233-b3a5-0030489edb7e for seqno 73280364 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 21:40.09 | ryan42 | but I can't figure it out. |
| 21:40.24 | [TK]D-Fender | <PROTECTED> |
| 21:40.34 | ryan42 | and I'm not sure if this line: -- Got SIP response 486 "Busy Here" back from 10.72.0.31:5090 .... could be a problem as welll |
| 21:40.41 | [TK]D-Fender | <robmal> You can view the screen of the phone via http. Very useful, but needs enabling after each restart. <- |
| 21:41.26 | robmal | [TK]D-Fender: http://community.polycom.com/t5/VoIP/FAQ-How-can-I-create-a-Screen-Capture-of-the-phone-GUI/td-p/4713 See the big red Note ;-) |
| 21:41.44 | [TK]D-Fender | ryan42, "sip set debug on" <- enable it and make a new PB so we can see what's actually happening. |
| 21:41.58 | robmal | You enable it via provision, but you need to run it every reboot. |
| 21:42.00 | ryan42 | okay. may take me a few moments to make it happen but i'll do that now |
| 21:42.45 | [TK]D-Fender | robmal, loh well |
| 21:43.26 | robmal | [TK]D-Fender: Still very useful, so i enable it by sending a combo of keys via push ;-) |
| 21:43.43 | ryan42 | [TK]D-Fender: is there anything in this I should filter before putting it online? |
| 21:43.50 | ryan42 | any credentials / etc? |
| 21:43.54 | [TK]D-Fender | no |
| 21:44.40 | ryan42 | ok |
| 21:45.26 | ryan42 | Here's the new paste: http://pastebin.com/jVU3XQYw |
| 21:46.03 | ryan42 | i did blank out the last 4 of the tel #'s to 0000 |
| 21:47.57 | [TK]D-Fender | ryan42, <--- Reliably Transmitting (no NAT) to 8.17.32.12:5060 --->Contact: <sip:9798360000@10.72.0.30:5061> o=root 907186043 907186043 IN IP4 10.72.0.30 |
| 21:48.14 | [TK]D-Fender | ryan42, You failed to set your system up to work properly from behind NAT |
| 21:48.54 | [TK]D-Fender | ryan42, "Asterisk SIP Settings" <---- fix your localnets, wan IP, etc, and prevent reinvites. |
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| 21:49.09 | [TK]D-Fender | ryan42, For any questions, take this up in #freepbx since you have to play by its rules |
| 21:49.20 | ryan42 | ahh. okay |
| 21:49.28 | ryan42 | Thank you! |
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| 21:49.57 | ryan42 | I'm on the fence about whether or not FreePBX was a good idea |
| 21:50.02 | ryan42 | but it does automate a lot of manual labor |
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