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| 00:24.15 | drab | looking through a ton of sip trunking providers and finding it hard to decide, is there a general consensous in the community (at least for US) on good providers? nextiva and 8x8 keep coming up and so does onsip |
| 00:24.54 | drab | however their pricing seem to be all pretty similar or obscure in a snse, I don't quite like for example that yuo need to request a quote to just see what it'd cost to rent a trunk |
| 00:25.19 | drab | on the other hand http://www.didforsale.com/products/sip-trunking-pricing this is pretty interesting |
| 00:25.32 | drab | quite a few diff plans and mixes with very clear prices, I tend to like that sort of transparency |
| 00:27.26 | drab | actually scratch onsip, doesn't seem to be doing trunking and let me run my own pbx |
| 00:28.19 | WIMPy | There is no such thing as a sip trunk. That's just Marketing gobbledigook. |
| 00:29.36 | drab | uhm, ok, one sip channel? |
| 00:30.19 | drab | ie make one call at a time, x "trunks" => x channels => x number of concurrent calls |
| 00:30.38 | mjordan | well, if you're in the lower 48, I'd say give Digium a try, but I'm obviously biased :-0 |
| 00:30.40 | mjordan | :-) |
| 00:30.48 | WIMPy | Beter talk about accounts. When you talk channel(s) that usually means the number of possible concurrent calls. |
| 00:33.11 | drab | ok, but aren't you really paying for those concurrent calls and DIDs? those seems to be what you're really buying (besides services if that's what you want) |
| 00:33.43 | WIMPy | Yes |
| 00:34.05 | WIMPy | But not all ITSPs have limits. |
| 00:36.02 | doop | drab: i've been happy with vitelity -- you may want to look at them |
| 00:39.47 | drab | doop: I looked at them, specifically http://www.vitelity.com/services_voip/ , but was confused by their pricing |
| 00:40.02 | drab | "4 Simultaneous incoming calls, No minute fees", how can I get 4 calls with just one "number"? |
| 00:40.32 | drab | like putting them on hold or something? but there's only one concurrent call going on at any time |
| 00:41.22 | WIMPy | Uh |
| 00:41.43 | WIMPy | There is no relation between numbers and the number of calls. |
| 00:43.06 | doop | you could have one number and 24 simultaneous calls to that number etc |
| 00:44.38 | drab | ok, I apologise, guess I'm confused and still thinking traditional phones, if I pick up and I'm talking to somebody else and another person tries to call they get a busy tone |
| 00:44.56 | doop | unless you have something called "call waiting" |
| 00:45.11 | drab | sure, but I'm still talking to only one person at a time (if it's not a conf call) |
| 00:45.14 | WIMPy | That still has nothing to do with havon one number. |
| 00:45.58 | WIMPy | Do you think that companies who advertise one phone numer can only talk to one customer at a time? |
| 00:46.06 | doop | when was the last time you saw someone tell you to call their 1-800 number and then call a different 1-800 number if that first one is busy? |
| 00:46.34 | mjordan | never? |
| 00:46.47 | mjordan | is pretty sure this is why he works on Asterisk |
| 00:47.02 | WIMPy | ? |
| 00:47.29 | WIMPy | To make call waiting possible in Asterisk? |
| 00:47.43 | mjordan | No, to make it so that if you dial a DID you can talk to more than a single person |
| 00:47.51 | mjordan | one DID != one device |
| 00:48.26 | WIMPy | is pretty sure we're talking about non-DID anyway. |
| 00:48.51 | doop | i think all directory numbers are called "DID" here anyway |
| 00:49.19 | WIMPy | Maybe, but it's wrong and doesn't really help. |
| 00:49.31 | drab | I think I got tripped by this whole misunderstanding around trunk and channel and the idea that I pay for one "trunk" and I can make only one call at a time |
| 00:49.36 | drab | anyway, thanks for your patience |
| 00:50.02 | WIMPy | "Trunk" in marketing usually means multiple calls. |
| 00:50.19 | drab | doop: so when vitelity talks about outbound terminations, is that where the channel thing kicks in? more than one termination => more than one outbound call at the same time? |
| 00:50.32 | doop | basically |
| 00:51.12 | WIMPy | mjordan: The funny thing is that I can still talk to more people without Asterisk. |
| 00:57.57 | carrar | !! |
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| 03:14.25 | monsterco | <PROTECTED> |
| 03:14.27 | monsterco | what is this? |
| 03:15.43 | WIMPy | a call from a call-file was completed. |
| 03:20.05 | monsterco | apparently maintenence polling for time conditions |
| 03:20.23 | monsterco | seems really labor intensive - not sure if I disable what the backdraw would be?? |
| 03:20.48 | WIMPy | What? |
| 03:32.09 | [TK]D-Fender | monsterco, Not a question for here |
| 03:32.14 | [TK]D-Fender | monsterco, And it's doing it for a REASON. |
| 03:32.44 | [TK]D-Fender | monsterco, do you start pulling the spark-plugs out of your car while driving on the highway? |
| 03:32.56 | [TK]D-Fender | monsterco, YES, there's a consequence |
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| 04:28.35 | carrar | This is why you install a spark-plug disconnect knife breaker switch in your car! |
| 04:28.56 | carrar | !! |
| 04:35.10 | [TK]D-Fender | IT'S ALIVE!!!! |
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| 04:38.08 | Jymmm | [TK]D-Fender: Got my hands on a ZTE WF720, working great =) |
| 04:38.27 | [TK]D-Fender | To fix your chan_mobile audio issues? |
| 04:38.58 | Jymmm | [TK]D-Fender: GSM-FXS (No VoIP) |
| 04:39.27 | [TK]D-Fender | ok... |
| 04:39.54 | Jymmm | Fixed Wireless (or Cellular) Terminal is what they're called. |
| 04:40.26 | [TK]D-Fender | So you plug an ANALOG phone into that, right? |
| 04:40.30 | Jymmm | Rare in North America, popular in Middle East, India, Australia it seems. |
| 04:40.33 | Jymmm | Correct. |
| 04:40.34 | [TK]D-Fender | and use as your "cell" |
| 04:40.42 | [TK]D-Fender | Ok..... |
| 04:40.48 | Jymmm | yes. |
| 04:41.18 | Jymmm | Has backup battery too |
| 04:41.27 | Jymmm | apx 1.5 hours |
| 04:42.15 | Jymmm | [TK]D-Fender: Has a TNC connector for the antenna, so you can use an external antenna too |
| 04:42.16 | [TK]D-Fender | And you're using a totally dumb analog phone with this? |
| 04:42.36 | Jymmm | Yeah, ATT Trim Phone (if you remember those =) |
| 04:42.52 | [TK]D-Fender | ok, have fun with it... |
| 04:43.02 | Jymmm | But, you canjust plug it directly into the house wiring too |
| 04:43.19 | Jymmm | and it'll ring all the phones |
| 04:43.41 | WIMPy | Hmm. Now what was the purpose of not just using a cellphone? |
| 04:44.05 | Jymmm | Loud ass ringer in the workshop |
| 04:44.29 | WIMPy | remembers |
| 04:44.40 | Jymmm | no aving to rememrb to recharge cellphones |
| 04:45.17 | Jymmm | But mostly external yagi antenna |
| 04:45.38 | WIMPy | is not a friend of battery operated devices for that reason. |
| 04:45.47 | Jymmm | about 90ft up a pine tree =) |
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| 04:48.34 | Jymmm | Not too shabby for a $15 investment |
| 04:49.33 | WIMPy | Using a digital connection with an analogue terminal feels bad. |
| 04:50.13 | Jymmm | Nah, nice blend of old school and modern |
| 04:51.13 | Jymmm | On a cordless plus the Trim phone. |
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| 05:41.03 | janicez | Nini |
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| 06:13.54 | ChannelZ | Huh.. $40 Grandstream SIP phone? I think that's the cheapest I've ever seen a phone |
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| 08:29.17 | tzafrir | Guess what this does: http://paste.debian.net/196486/ |
| 08:32.22 | Tim_Toady | lol |
| 08:32.44 | Tim_Toady | its main purpose is to ensure job security :P |
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| 08:49.51 | tzafrir | Tim_Toady, no. It's a feature (intended for testing) |
| 08:50.10 | tzafrir | a multi-leged PRI call. |
| 08:50.31 | tzafrir | It takes a surisingly short time to establish |
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| 10:42.12 | janicez | Mew |
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| 12:00.24 | wasanzy | hi |
| 12:01.39 | wasanzy | I have added new music on hold class in /etc/asterisk/musiconhold.conf and add it to my queue config like this: musicclass=bima but the default is still playing after reloading the core |
| 12:02.16 | robmal | moh show classes |
| 12:02.21 | robmal | moh show files |
| 12:02.53 | wasanzy | Class: default |
| 12:02.53 | wasanzy | Mode: files |
| 12:02.54 | wasanzy | Directory: moh |
| 12:03.06 | wasanzy | looks like only the default kept showing |
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| 12:07.12 | wasanzy | http://pastebin.com/cnMYQ5mx this is my musiconhold.conf |
| 12:07.41 | wasanzy | do I need to run special command to reload the config other than core reload? |
| 12:09.23 | robmal | moh reload ? |
| 12:10.04 | wasanzy | I did that so many times but the new class is not showing |
| 12:10.24 | robmal | :-( |
| 12:13.02 | wasanzy | is there something am not doing well? |
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| 12:14.48 | robmal | I'd guess permissions. |
| 12:17.27 | wasanzy | -rw-r--r-- 1 asteriskpbx asteriskpbx 346 Jun 3 11:22 music_on_hold.alaw |
| 12:19.18 | robmal | Move the bima dir outside moh, maybe that's the problem. |
| 12:21.48 | wasanzy | ok |
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| 12:23.59 | wasanzy | worked now |
| 12:24.04 | wasanzy | thank you |
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| 12:37.16 | tenspeed705 | \quit |
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| 12:57.32 | tenspeed705 | anyone here using Grandstream phones? We launched a few in to testing, and seem to be working pretty well. 1 minor issue, is a delay when answering a call. The first few RTP packets, are not read by * properly and I get back RTP to short errors. |
| 12:58.07 | [TK]D-Fender | ~grandstream |
| 12:58.07 | infobot | grandstream is probably the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 12:58.10 | [TK]D-Fender | ~gs |
| 12:58.10 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
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| 12:58.55 | tenspeed705 | but but it runs Android lol |
| 12:59.50 | iulhk | using asterisk 13, caller making audio call, at receiver-end video screen automatically opened? |
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| 13:00.38 | tenspeed705 | honestly, I did not like GS years back, They seemed to have stepped up the game, seem to be working rather well this time around |
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| 13:26.10 | monsterco | I have a Siemens OpenStage 60 but I can't find proxy registeration - could it be locked or provisioned somehow? |
| 13:26.54 | robmal | My favourite phones! |
| 13:27.27 | robmal | Did you read the administration manual? |
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| 13:32.42 | [TK]D-Fender | robmal: #rhetorical |
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| 13:35.33 | robmal | That manual has something about 300 pages and it's worse than anything that you have ever seen, i asked just out of curiosity ;-) |
| 13:36.29 | robmal | But i managed to make a provisioning system for those phones! Never again. |
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| 13:57.15 | WIMPy | wasanzy: Next time do 'module reload res_musiconhold' |
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| 14:36.41 | cervajs2 | hello, can you point me to some examples for asterisk+statsd in asterisk13 ? i dont understand how send data (https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_statsd) |
| 14:38.39 | mjordan | cervajs2: right now, the only data is sent is sent automatically by the res_chan_stats module |
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| 14:39.10 | mjordan | the res_statsd module just configures which statsd server receives the data sent by the res_chan_stats module. |
| 14:40.19 | hayonj | can anyone shed some light about how to resolve DNS issues when using just a static address but we didn't purchase a hostname or an A or AAAA record |
| 14:41.50 | cervajs2 | mjordan: thanks. i found http://www.slideshare.net/danjenkins/bringing-realtime-voip-metrics-to-2013 it's really great presentation |
| 14:42.34 | mjordan | yeah, Dan's pretty awesome :-) |
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| 14:53.02 | cervajs2 | mjordan: can you public your dangerous demo from berlin? i voted for you ;) |
| 14:58.45 | mjordan | cervajs2: sure, I can put it up on github a bit later in the day. |
| 15:00.28 | cervajs2 | mjordan: thanks. i'll check your blog tomorrow |
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| 15:25.23 | cervajs2 | my webrtc calls ends after ~60seconds with "res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3 alert unexpected message', terminating". any ideas where can be problem? |
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| 16:31.40 | anonymouz666 | hello |
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| 16:37.45 | xochilpili | hello everybody! |
| 16:38.56 | ChannelZ | OH HAI |
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| 17:36.46 | tenspeed705 | Hey guys, Odd request, curious if anyone may have a pointer. I have a customer that wants to have 3 voicemail greetings. a day time, a night and a weekend. Anyone have a way I can do this, without using more 2 or 3 mailboxes? |
| 17:40.12 | rjvvliet | tenspeed705 : Maybe use a playback with gotiftime and disable the voicemailvox greeting? |
| 17:41.04 | [TK]D-Fender | yup. |
| 17:41.37 | [TK]D-Fender | Of do a time check and directly swap the files before calling VoiceMail |
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| 17:42.16 | tenspeed705 | how would I disable the greeting, is that an option in app voiemail? |
| 17:42.57 | [TK]D-Fender | yes |
| 17:43.10 | [TK]D-Fender | Reading the instructions is highly recommended |
| 17:43.22 | carrar | crazy talk!! :) |
| 17:43.31 | rjvvliet | core show application voicemail |
| 17:44.05 | tenspeed705 | I read that, I think on first read, I think I mistaken what the s option does |
| 17:45.08 | rjvvliet | o maybe just an empty greeting and relay on the PlayBack() |
| 17:49.20 | tenspeed705 | Thanks Guys, was my mistake, I missunderstood what would happen if I used just the s option |
| 17:54.17 | leftleg | hello all |
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| 18:07.39 | qakhan | anyone setup Polycom soundpoint 670 auto answer? |
| 18:11.25 | qakhan | i have config it and it is working, but if user is already on call (non auto answer line) and other call comes for auto answer line, user can see auto answer call on phone. after 1 min phone put current call on hold and auto answer 2nd call. |
| 18:13.47 | robmal | Set calls per line to 1 |
| 18:14.38 | *** join/#asterisk averythomas (~averythom@2607:5300:60:2d42::1) |
| 18:14.47 | qakhan | in phone? |
| 18:14.53 | robmal | Yes. |
| 18:16.27 | qakhan | which .cfg? |
| 18:17.10 | robmal | Geez. |
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| 18:29.24 | qakhan | robmal i set call per line to 1 |
| 18:29.28 | qakhan | but same issue |
| 18:33.35 | robmal | Not possible. |
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| 18:35.39 | qakhan | let me repeat my question. |
| 18:35.41 | *** join/#asterisk reveal (32c4e3fe@gateway/web/freenode/ip.50.196.227.254) |
| 18:36.05 | qakhan | i have Polycom soundponint 670 |
| 18:36.18 | reveal | can anyone recommend an alternative to Sangoma A102X PCI-Express cards that are compatible with asterisk |
| 18:36.20 | qakhan | 3288 registered on line 1 |
| 18:36.31 | qakhan | which is standard line |
| 18:38.36 | qakhan | 5801 on line 2 (which is auto answer) line |
| 18:40.08 | qakhan | when 3288 line 1 on call, and a call comes on 5801 line 2. it shows on phone and after 1 min phone put 3288 on hold and answer 5801 |
| 18:40.46 | danielbw | okay |
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| 18:43.14 | qakhan | how to fix it? |
| 18:46.50 | reveal | well line 2 is auto answer isnt it doing its job |
| 18:47.42 | qakhan | but it should check if phone is aleard on call or not |
| 18:47.45 | qakhan | roght? |
| 18:48.26 | robmal | It's auto answer so no :-) |
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| 18:49.19 | reveal | its job is to take calls coming in and auto answer them not check them if another line is open |
| 18:50.02 | reveal | why not configure a loop or use a hunt group |
| 18:50.13 | qakhan | ok so what is the sulotion for that |
| 18:51.01 | robmal | Hmm. |
| 18:51.19 | robmal | Note: Using the Answer Ring Type |
| 18:51.19 | robmal | The auto-answer on incoming call is currently only applied if there is no other call in progress on the phone at the time. |
| 18:51.21 | reveal | are you using a crm or |
| 18:51.33 | robmal | In UC 4.0.5 it shouldn't pick up the new call. |
| 18:51.46 | reveal | i guess the question is which version hes using |
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| 18:55.09 | Kunsi | hello, ihave a POTS line set up in my asterisk setup, which currently just uses Goto() to go to my local extension 76, which dials one of my phones. now i want to modify it to also dial local extension 91, which is another phone - could be done with Dial(SIP/76&SIP/91). but i want to be able to pick up tha call using Pickup(76) OR Pickup(91) - is that possible to do? |
| 18:56.57 | qakhan | robmal i am using UC Software Version 4.0.8.1608 |
| 18:57.14 | robmal | My favourite. |
| 18:57.27 | robmal | So, show us the configs. |
| 18:57.54 | reveal | kunsi, should be possible |
| 18:58.50 | Kunsi | any tips on what to look for at the docs? |
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| 19:01.29 | reveal | kunsi, what about creating a pickup group |
| 19:01.39 | reveal | http://edoceo.com/exemplar/asterisk-call-groups |
| 19:01.43 | Kunsi | yeah, found that like a second ago |
| 19:01.51 | reveal | hehe |
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| 19:02.25 | iulhk | using asterisk 13 , caller making audio call, at receiver-end after accepting call, video windows automatically on, how to fix it? any hint pls? |
| 19:04.09 | reveal | could be a conf setting? |
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| 19:15.10 | qakhan | robmal which config you want to see? |
| 19:20.43 | robmal | <se /> part |
| 19:23.36 | qakhan | here is my complete sip.cfg |
| 19:23.37 | qakhan | http://pastebin.com/NkKkn4Ng |
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| 19:27.54 | [TK]D-Fender | qakhan: no point |
| 19:27.56 | [TK]D-Fender | 'qakhani have config it and it is working, but if user is already on call (non auto answer line) and other call comes for auto answer line, user can see auto answer call on phone. after 1 min phone put current call on hold and auto answer 2nd call. |
| 19:28.04 | [TK]D-Fender | qakhan: If they are on a call it will NEVER auto-answer |
| 19:28.11 | [TK]D-Fender | It will NEVER interrupt someone on a call |
| 19:28.24 | [TK]D-Fender | It'll just ring like any other call |
| 19:28.46 | reveal | qakhan, what about http://pastebin.com/FsW05FHa |
| 19:29.40 | qakhan | robmal was asking about polycom sip.cfg |
| 19:30.10 | qakhan | [TK]D-Fender i was think same way, it should not asnwer the call. |
| 19:31.12 | qakhan | here is 1 thing. both exts are on different asterisk. 3288 is on Asterisk A and 5801 is on Asterisk B. |
| 19:31.21 | robmal | qakhan: Try changing it like reveal suggested. |
| 19:31.34 | [TK]D-Fender | Doesn't matter |
| 19:31.39 | [TK]D-Fender | the PHONE will not interrupts you |
| 19:31.54 | [TK]D-Fender | It doesn't care what other call you are on. |
| 19:32.02 | [TK]D-Fender | If you are one it will not interrupt that call |
| 19:33.03 | WIMPy | wonders what else it could do? |
| 19:33.04 | qakhan | [TK]D-Fender is it answering the call after 1 min |
| 19:33.24 | WIMPy | And what does it do at that time? |
| 19:33.25 | cervajs2 | my webrtc calls ends after ~60seconds with "res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3 alert unexpected message', terminating". any ideas where can be problem? |
| 19:33.26 | qakhan | reveal what did you suggest? |
| 19:33.51 | qakhan | it put line 1 on hold and answer the line 2 |
| 19:34.19 | WIMPy | Your active call is put on hold automatically? |
| 19:34.39 | qakhan | yes |
| 19:34.51 | WIMPy | That's evil. |
| 19:36.12 | qakhan | yes it is |
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| 19:41.39 | qakhan | guys any thought? |
| 19:43.48 | WIMPy | File a bug report. |
| 19:44.07 | robmal | Did you try adding se.rt.ringAutoAnswer like reveal suggested? |
| 19:50.16 | leftleg | anyone ever use the xact dialer for freepbx? |
| 19:51.03 | WIMPy | I'd assume that who ever wrote it does so. |
| 19:51.39 | qakhan | robmal what is complete se.rt.ringAutoAnswer setting? |
| 19:52.00 | robmal | 21:29:32 < reveal> qakhan, what about http://pastebin.com/FsW05FHa |
| 19:55.03 | qakhan | robmal do i remove my previous config? overwirite with this? |
| 19:55.42 | robmal | Only the <se /> part, leave <voIpProt /> as is. |
| 19:56.10 | robmal | Oh, i just noticed you are missing <se> and </se> around <se.rt /> |
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| 19:58.37 | qakhan | ok thanks, let me check with this |
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| 20:14.36 | qakhan | robmal i have tested it but same result |
| 20:15.28 | robmal | Meh. |
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| 20:16.54 | robmal | Well, only option now is to go to community.polycom.com and open a new topic. I hope Steffen will figure this out. |
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| 20:19.44 | qakhan | ok Thanks, i already opened a topic |
| 20:22.23 | *** part/#asterisk digitalirony (~digitalir@magrathea.milkyway.the-galaxy.net) |
| 20:33.46 | zmitya | hi guys |
| 20:34.30 | zmitya | one of my friend lost his asterisk box, I would like to help him installing it but we have issues with the ISDN card... could somebody please help us ? |
| 20:35.05 | WIMPy | ~ask |
| 20:35.05 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:35.54 | zmitya | well, I whish I could ask better .. :( |
| 20:36.20 | zmitya | we have a Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] card.. we should set up it on a Debian.. |
| 20:36.31 | WIMPy | then tell us what you have and what you want. |
| 20:36.52 | zmitya | I have DAHDI installed |
| 20:37.02 | zmitya | there is a card which works well |
| 20:37.24 | zmitya | but this above mentioned ISDN card does not work |
| 20:37.38 | zmitya | I installed mISDN this way: https://www.misdn.eu/wiki/Howto_for_Debian/ |
| 20:38.24 | WIMPy | You need to decide if you want to use mISDN or DAHDI. You can't use both. |
| 20:38.29 | WIMPy | (not with one card that is) |
| 20:38.31 | zmitya | misdn_info shows this: |
| 20:38.33 | zmitya | http://pastebin.com/0KgncVCy |
| 20:38.51 | zmitya | WIMPy: ok, so thats a new info to me :( |
| 20:39.10 | zmitya | I have DAHDI installed, and there is a card which works well |
| 20:39.12 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 20:39.31 | zmitya | there are 3 input lines, 2 goes to the "bigger" card, that works with DAHDI |
| 20:40.03 | WIMPy | Err, I don't understand that. |
| 20:40.14 | zmitya | there is a "smaller" card the above one that gets an RJ45 cable |
| 20:40.26 | zmitya | that is the 3rd cable.. |
| 20:40.34 | zmitya | so this one does not work.. |
| 20:40.53 | WIMPy | Are you talking about two Asterisk boxes? |
| 20:41.11 | zmitya | WIMPy: no, I have one.. That one has 2 ISDN cards |
| 20:41.19 | zmitya | WIMPy: ^ |
| 20:41.39 | WIMPy | So a 4 and a single port card? |
| 20:41.52 | zmitya | yes, exactly |
| 20:41.55 | WIMPy | And the 4 port one uses DAHDI? |
| 20:41.56 | zmitya | the 4 port card works |
| 20:42.02 | zmitya | exactly |
| 20:42.17 | zmitya | I got that working, I get incmoing calls on it.. |
| 20:42.29 | WIMPy | Well, might make sense to use the same stack for both. |
| 20:42.46 | zmitya | WIMPy: ok |
| 20:43.01 | zmitya | but no idea how to do that |
| 20:43.19 | WIMPy | Get the dahdi_hfcs module. |
| 20:43.46 | zmitya | I installed DAHDI by this: |
| 20:43.50 | zmitya | http://paulswasteland.blogspot.hu/2013/10/quickly-install-asterisk-and-dahdi-on.html |
| 20:44.20 | WIMPy | That module is not part of DAHDI. |
| 20:45.07 | zmitya | hm |
| 20:45.24 | WIMPy | git clone git://dahdi-hfcs.git.sourceforge.net/gitroot/dahdi-hfcs/dahdi-hfcs |
| 20:45.31 | WIMPy | That's the latest I know. |
| 20:46.58 | zmitya | whill this interfere with my dahdi ? |
| 20:47.13 | zmitya | WIMPy: or is it different ? |
| 20:47.24 | WIMPy | It should. |
| 20:47.41 | WIMPy | But you never know if things become incompatible in later versions. |
| 20:47.51 | WIMPy | It has been a long time since I tried. |
| 20:48.20 | WIMPy | I usually use mISDN and LCR. |
| 20:50.47 | zmitya | WIMPy: ok, let me give a try |
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| 20:53.55 | zmitya | WIMPy: I should play with mISDN 1.x right ? |
| 20:54.02 | zmitya | not with mISDN2 ? |
| 20:54.41 | WIMPy | I don't know if you can still get the old version running. |
| 20:55.05 | WIMPy | But for Asterisks chan_misdn you need v1. V2 goes with LCR. |
| 20:55.39 | zmitya | so which way would you recommend ? |
| 20:56.03 | WIMPy | Depends on what features you want. They are all different. |
| 20:56.22 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/3-3 |
| 20:57.21 | zmitya | WIMPy: well, these features does not tell me anything :( I am not a telephony engineer :( |
| 20:57.57 | zmitya | I am just helping my friend out to build recover his * box to get incoming calls tomorrow.. |
| 20:58.22 | WIMPy | Do yu only connect lines or do you also want to connect phones? |
| 20:58.46 | zmitya | we have a few SIP clients connected to * |
| 20:58.52 | zmitya | so * has only lines |
| 20:58.58 | WIMPy | To the cards. |
| 20:59.01 | zmitya | yes |
| 20:59.22 | WIMPy | Bad luck, you can do that with all of them. |
| 20:59.36 | zmitya | well, thats good :) |
| 21:00.03 | WIMPy | You still got 3 ways to choose from. |
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| 21:07.33 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 21:07.35 | xochilpili | hi al |
| 21:08.14 | xochilpili | i have a fresh install of asterisk now , but im trying to load ooh323 module but i got an error: undefined symbol: ast_rtp_instance_get_incoming_source_address |
| 21:08.44 | xochilpili | installed from yum asterisk13-addons-ooh323.x86_64 |
| 21:08.55 | xochilpili | is there a way to fix this? |
| 21:08.58 | xochilpili | thanks |
| 21:11.13 | zmitya | WIMPy: do you know why this server has 2 ISDN cards ?? can I use the 3rd and 4th socket in the "big" card somehow ? |
| 21:11.34 | WIMPy | Sure. |
| 21:11.47 | zmitya | the single port gets an RJ45, but the 2 others are smaller |
| 21:11.53 | WIMPy | Unless it's only a 2 port card. |
| 21:12.03 | WIMPy | Huh? |
| 21:12.04 | zmitya | no its a 4 port card |
| 21:12.24 | WIMPy | Are you sure the other card is ISDN? |
| 21:12.41 | zmitya | hm, no, let me check what it is |
| 21:13.33 | zmitya | WIMPy: 03:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 21:13.45 | xochilpili | anyone? |
| 21:14.17 | WIMPy | That's unfortunately useless. That PCI ID is known to be in use by different kind of el cheapo cards. |
| 21:14.46 | WIMPy | Try to look at - err - dahdi_scan? dahdi_hardware? |
| 21:15.20 | zmitya | WIMPy: ok, it is now rebooting, a few secs.. |
| 21:18.30 | zmitya | WIMPy: http://pastebin.com/g3VYRSKk |
| 21:19.07 | WIMPy | Ok, so that is POTS. |
| 21:19.44 | zmitya | this means that I can't hack ISDN line there right ? :) |
| 21:19.47 | zmitya | WIMPy: ^ |
| 21:19.53 | WIMPy | Why do you still use POTS if you also have ISDN? |
| 21:19.58 | WIMPy | yes |
| 21:20.20 | zmitya | WIMPy: yes, they use all 3 lines |
| 21:21.09 | zmitya | WIMPy: look at this please: http://pastebin.com/Rkdd3caM |
| 21:21.21 | zmitya | it sees both cards |
| 21:21.46 | zmitya | do you think that this dahdi could get the other working as well ? |
| 21:21.55 | zmitya | WIMPy: ^ ?? |
| 21:24.17 | WIMPy | Doesn't look bad. |
| 21:24.36 | *** join/#asterisk acidfoo (~netvirt@unaffiliated/acidmen) |
| 21:26.19 | acidfoo | hi ! is it normal that if I hangup while Record() is going on, the file gets delete ? ( Asterisk 11.17.0 ) my dialplan is a simple Record(file.wav,0,60,a) followed by hangup() |
| 21:26.27 | *** join/#asterisk madax (~madax@c3-167.i07-1.onvol.net) |
| 21:28.11 | acidfoo | or how can I prevent this ;) |
| 21:28.41 | WIMPy | 'core show application Record' |
| 21:29.45 | acidfoo | sounds like that should do the trick ! k: Keep recorded file upon hangup. |
| 21:29.45 | madax | Hi guys, does anyone knows if it is possible to dynamically plays a sound file to a caller or/and a callee please ? (I'm using asterisk Lts 11.17.1) |
| 21:30.03 | matt_ | does anybody know if 'itslenny' or an alternative is avaliable? or just recordings from it? |
| 21:35.42 | *** part/#asterisk mjordan (mjordan@nat/digium/x-lfcmysbbosbtcqju) |
| 21:39.48 | [TK]D-Fender | acidfoo, That protects against the actual CHANNEL hanging up. You dialplan app called afterwards has no impact on this. |
| 21:40.06 | [TK]D-Fender | acidfoo, that is, while in process of recording |
| 21:40.54 | acidfoo | since I append every recorded message to the same file, I don't like it to be deleted :) |
| 21:41.04 | acidfoo | so yeah I tested with the option 'k' and my file is not |
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| 22:49.00 | danielbw | Sangoma tech support has confirmed that the rx audio before and after the echo cancellation is crystal clear. Asterisk is somehow taking a crystal clear input from my Sangoma A101D and sending an rtp stream with wrong timestamps and interrupted talking with silence. |
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| 23:44.33 | danielbw | so at this point i need to troubleshoot whatever is causing the altered audio |
| 23:46.28 | danielbw | Does the jitter buffer parameters in chan_dahdi.conf affect anything on the rx side of a call coming from our PSTN though the A101D? |
| 23:50.29 | danielbw | Also, is this odd? * I call my from me cell phone. * Call comes into asterisk and rings my extension (SFLPhone 1.3.0). * I double click to answer. * I hear ringing on my SFLPhone. * I speak on my Cell Phone and the ringing stops. |
| 23:51.45 | danielbw | s/I call my from me cell phone/I call from my cell phone |
| 23:59.36 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |