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| 00:35.46 | tessier | Anyone had any problems with latency on the PSTN side of Teliax? I have been using them for years with no problem but suddenly we are occasionally seeing as much as 5s latency in calls. I cannot reproduce it. 22ms from my phone to my asterisk box and another 3ms to lax.teliax.net with zero packet loss. |
| 00:36.06 | tessier | Teliax has done a packet trace on a call during which you could obviously hear latency and proclaimed it real-time with no packet loss. They could not explain it. |
| 00:36.11 | tessier | station-to-station calls always seem good. |
| 00:36.53 | tessier | Some here have reported hearing "static". Given that this is digital end to end I have to wonder... |
| 00:44.37 | tessier | Wow...sure has gotten quiet in here these days. I remember back in 2005 or so this channel was constant scrollage. |
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| 00:48.34 | russellb | tessier: well this particular week was astricon ... many have been busy with that. |
| 00:56.16 | Juggie | russellb, back @ home? |
| 00:56.34 | russellb | nope ... in flight |
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| 01:00.18 | tessier | russellb: You are on an airplane right now? Awesome. |
| 01:00.28 | tessier | I have not yet had the pleasure of in-air wifi despite all of my travel. |
| 01:00.48 | tessier | Warflying doesn't count since we didn't transmit. |
| 01:01.22 | tessier | Has anyone ever heard of PSTN-induced latency? |
| 01:01.31 | tessier | Could Teliax just have a bad line card or something and not be aware of it? |
| 01:01.53 | WIMPy | Only with NGN "PSTN". |
| 01:01.57 | tessier | NGN? |
| 01:02.14 | WIMPy | Next Generation Network |
| 01:02.16 | tessier | Packet traces between my asterisk and teliax show real time no packet loss. |
| 01:02.22 | tessier | WIMPy: Ah. Right. |
| 01:02.29 | tessier | NGN PSTN aka VOIP. ;) |
| 01:02.38 | WIMPy | Yes. |
| 01:03.10 | tessier | This is why I am at my witts end with this one. Teliax recorded the packets. Played it back with me on the phone listening. We could clearly hear a 5 second delay between the speakers. |
| 01:03.15 | tessier | But the packet trace was perfect. |
| 01:03.17 | WIMPy | Delays of 3½ or 7 seconds were quite common whe I had a NGN line form a provider using CCM. |
| 01:03.33 | tessier | I am wondering if they aren't doing some sort of VOIP backhaul. Even though they swear they aren't. |
| 01:03.42 | Juggie | russellb, hah wifi in flight, nice. |
| 01:04.26 | tessier | I've sent them thousands of dollars in business. I am going to call them again tomorrow and get some more help with this. Because this just doesn't make sense. |
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| 05:38.58 | geneg1 | Is there an option for the Dial command such that it will ring for n seconds even if the destination channel is not available? |
| 05:39.24 | geneg1 | For example, if a softphone is not connected to an extension, the line should still ring for 20 secs before going to voicemail. |
| 05:39.38 | geneg1 | Default behaviour is that the call will go straight to v-mail if the softphone doesn't have a current registration. |
| 05:40.38 | WIMPy | Not that I know of, but you could add a local cahnnel that Wait()s. |
| 05:41.19 | geneg1 | Which conf file would I put such a channel in? iax.conf perhaps or channels.conf? |
| 05:41.54 | WIMPy | local channels are channels to the dialplan. |
| 05:42.08 | geneg1 | extensions.conf then? |
| 05:42.31 | WIMPy | yes |
| 05:42.44 | geneg1 | WIMPy: thanks for the tip, that gives me something to start with. |
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| 05:54.52 | MCIML | hey guys im following the new starfish for 1.8 and i'm doing the configuring over asteriskNOW install |
| 05:55.20 | MCIML | i got to the part where i'm trying to modify asterisk.conf and i cant seem to edit the file, permission denied |
| 05:59.52 | MCIML | nevermind did chown -R on dir and that worked |
| 06:01.23 | geneg1 | WIMPy: worked perfectly, thanks again. |
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| 06:57.19 | v1s | I have 2 different providers. what is the best way to use them if one fails in dialing to try the next? Like should I just do Dial(SIP/PROV1&SIP/PROV2) |
| 06:59.13 | v1s | or should I use ChanIsAvail ? |
| 06:59.16 | [TK]D-Fender | v1s: No, that calls them both at the same time |
| 06:59.34 | [TK]D-Fender | v1Just dial them back to back |
| 07:00.07 | v1s | u mean just dial prov1 then prov2 right after? |
| 07:00.21 | [TK]D-Fender | yes |
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| 07:02.44 | v1s | thanks |
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| 07:07.19 | TehRabbitt | anyone here know if it's possible to set asterisk up to send and receive faxes via email? like assign a DID just for faxes and receive them to email? |
| 07:08.27 | TehRabbitt | or would I be better off just connecting a Fax machine to my linksys PAP2, and configure it as a line? |
| 07:09.20 | TehRabbitt | Anyone here tonight? |
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| 07:13.44 | jdoe | heh, just out of curiosity, did you even try google? |
| 07:14.16 | jdoe | it's the first hit for every combination of "asterisk", "fax" and "email" I can think of. |
| 07:14.19 | TehRabbitt | jdoe: yes, but the only solution I found is one that relies on openoffice |
| 07:14.34 | jdoe | ... what? |
| 07:14.54 | TehRabbitt | http://asterfax.sourceforge.net/ |
| 07:15.13 | jdoe | ... yeah I dunno, in any case, http://www.voip-info.org/wiki/view/Asterisk+Fax+to+email is the first hit here. |
| 07:15.20 | TehRabbitt | ah |
| 07:15.34 | jdoe | haven't done it personally, it likely works though. |
| 07:15.39 | jdoe | or at least, likely a starting point. |
| 07:15.46 | TehRabbitt | i'm guessing my best bet is to buy a seperate DID for the fax as well? heh |
| 07:16.29 | jdoe | I would think, yeah. |
| 07:16.50 | jdoe | maybe you can share one, I don't know. |
| 07:17.17 | jdoe | it was on my todo list, then we figured it was easier to just get a pstn line for fax. |
| 07:17.30 | TehRabbitt | ah |
| 07:17.33 | jdoe | er |
| 07:17.36 | jdoe | sorry |
| 07:17.55 | TehRabbitt | yea idk, I don't have a fax machine, and idk where i'd put one 0_o though I do have a flatbed scanner so I figure that's better than nothing lol |
| 07:17.59 | jdoe | I should clarify. I'm not discouraging this way, I just had higher priority things to do. |
| 07:18.38 | TehRabbitt | i mean this isn't top priority, but having to go down to my local library to fax things at 1.50/page, it's annoying after the first 30 pages :-\ |
| 07:18.40 | TehRabbitt | lol |
| 07:18.59 | jdoe | meh. dids are cheap. |
| 07:19.18 | TehRabbitt | i figure even if I send one fax a year, it's cheaper than the library |
| 07:19.18 | TehRabbitt | lol |
| 07:19.40 | TehRabbitt | i mean i'm using flowroute right now for my DID but i'm pretty sure there are cheaper ones, no? |
| 07:20.13 | jdoe | I don't know anything about them, sorry |
| 07:28.29 | TehRabbitt | I have a feeling someone here is going to say "read the book" for this question but I wanted to just ask how hard it is to accomplish this... I want to have in my voice menu, an option for "Press 1 if you are located in XYZ county, press 2 if you are located in YZX county" etc... and then drop them to another menu that says "For tech support press 1, etc" |
| 07:32.18 | [TK]D-Fender | TehRabbitt: under a dozen lines |
| 07:32.38 | TehRabbitt | i'm guessing just use gotos? |
| 07:33.10 | [TK]D-Fender | TehRabbitt: Each IVR would be in its own context, so yes, Goto() between them as you go deeper, etc |
| 07:33.47 | TehRabbitt | Ah, so basically make a context for each level, and then just logically map it out so one level goes to the next? |
| 07:33.51 | [TK]D-Fender | [03:07]<TehRabbitt>anyone here know if it's possible to set asterisk up to send and receive faxes via email? like assign a DID just for faxes and receive them to email? <- ReceiveFax() + call Sendmail YOURSELF to send it |
| 07:34.01 | [TK]D-Fender | TehRabbitt: yes |
| 07:35.10 | TehRabbitt | TKD-Fender: In regard to the fax thing... basically even if it were to just save the faxes into a directory that I can retrieve them from that'd be good enough... and is there a way to use my existing DID or would I be better off getting a seperate DID just for faxes? |
| 07:35.24 | [TK]D-Fender | TehRabbitt: Separate |
| 07:35.41 | [TK]D-Fender | TehRabbitt: And you'd better be with a provider supporting T.38 |
| 07:35.45 | TehRabbitt | ah |
| 07:36.05 | rdahlin_1 | Has anyone more than mine PBX been "hit" by a botnet in the last week ??? |
| 07:38.06 | [TK]D-Fender | rdahlin_1: Of course. |
| 07:38.18 | [TK]D-Fender | rdahlin_1: Happens all the time |
| 07:38.26 | TehRabbitt | what do you mean happens all the time? 0_o |
| 07:38.53 | [TK]D-Fender | just what it sounds like |
| 07:39.04 | TehRabbitt | whats the point of the botnets though? |
| 07:39.14 | rdahlin_1 | Fender: OK. I have not seen anything before this week and I my PBX has been online for at least 4-5 years.... maybe I just have been lucky.... |
| 07:40.12 | [TK]D-Fender | rdahlin_1: http://www.despair.com/trouble.html |
| 07:40.13 | rdahlin_1 | TehR: When I look in my logs they are just trying to "bruteforce" usernames from 0 - 10000 and some random usernames |
| 07:40.35 | rdahlin_1 | Fender: ahhh :) |
| 07:40.49 | rdahlin_1 | :( |
| 07:41.05 | rdahlin_1 | http://kb.smartvox.co.uk/index.php/asterisk/secure-asterisk-pbx-part-1/ |
| 07:42.28 | [TK]D-Fender | bedime, back later. |
| 07:42.32 | rdahlin_1 | Fender: do you know if there is any way to block an IP after say, 10-20 bad attempts in one second..... |
| 07:42.46 | rdahlin_1 | bye (I just woke up.... 10 AM here ;) ) |
| 07:42.51 | Marquel | rdahlin_1: how about iptables recent ;) |
| 07:43.37 | rdahlin_1 | Marquel: hmmm PM ? |
| 07:44.52 | Marquel | rdahlin_1: using recent target for brute force attacks against my sshd ;) |
| 07:45.22 | rdahlin_1 | ok.... |
| 07:46.14 | Marquel | rdahlin_1: i just don't know how good it'll work on udp-based traffic. |
| 07:47.21 | rdahlin_1 | mhmmm... I'll talk to a friend about this... he's kind a network guru... ;) |
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| 08:02.43 | ectospasm | rdahlin_1: you may want to look into something like fail2ban, not sure if it can be adapted for SIP connections. |
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| 08:36.01 | rdahlin_1 | ectospasm: thanx..... |
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| 09:51.11 | dandate2 | when configuing a multi-line ATA device does each line need to use a different sip port i.e. line1=5060, line2-5061 etc. |
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| 12:56.24 | vk4akp | Hi guys! |
| 12:56.44 | vk4akp | How hard is it to add voice mail to my TDM400P extension so after it rings out people can leave a message? |
| 12:59.12 | Gugge | put a Voicemail() after the Dial() command in the dialplan ? |
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| 12:59.34 | vk4akp | Is that all? |
| 12:59.43 | vk4akp | I don't need to set up lots of stuff? |
| 12:59.54 | vk4akp | How about getting it forwarded to me via email as well? |
| 13:00.00 | Gugge | you need to setup voicemail.conf |
| 13:00.08 | Gugge | for voicemail to work |
| 13:00.16 | vk4akp | I have a demo voicemail.conf there already from teh initial install. |
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| 13:00.36 | Gugge | then just edit that to do what you like :) |
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| 13:14.08 | vk4akp | ok Thanks. |
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| 13:14.24 | vk4akp | To send it to email though do I need an email server on my Box? |
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| 14:08.13 | jeffik | ariel_: Hi, are you in California? |
| 14:27.54 | ariel_ | hi jeffik no I am not |
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| 15:04.43 | amdpower | Anybody have any luck getting sdp passing video params for videosupport? It defaults to 5 gps for most of my endpoints. |
| 15:05.01 | amdpower | Fps that is. |
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| 15:51.09 | stromboli | anyone in here familiar with chan_skinny ? |
| 15:52.10 | beardy | Familliar as in name. But didn't bother with it. I run SIP. But you'll have more success asking a real question. |
| 15:52.22 | stromboli | ok |
| 15:53.21 | *** join/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 15:53.52 | beardy | :) |
| 15:54.17 | stromboli | Well, i had a skinny setup with my 7960's and 7970 using Asterisk 1.6 and I recently compiled asterisk 1.8 and my skinny.conf stopped working, so i re confgured with a default template and i cant get skinny to work... it keeps complaining about you need at least one line on a device |
| 15:54.38 | stromboli | and i have two lines setup in the device in the conf file |
| 15:55.42 | gregd | guys, could someone help me with configuring trunk to use spa3102 pstn? I thought I did everything right, however on my spa3102, i've got Info->Pstn line status->registration state set to Failed.. |
| 15:55.49 | stromboli | it also seems that they took a lot of the device conf options out for Skinny |
| 15:56.32 | stromboli | Oct 30 11:56:05] ERROR[29179]: chan_skinny.c:7090 config_device: A Skinny device must have at least one line! |
| 16:01.13 | beardy | stromboli: As I understand it the skinny driver doesn't support as much things as the not in Asterisk, SCCP driver? Have you considered running the SIP image instead? (unrelated I know) Has the name of the config file changed? |
| 16:01.44 | *** join/#asterisk [cannibalera] (~cannibale@201-41-247-4.fnsce703.dsl.brasiltelecom.net.br) |
| 16:01.57 | stromboli | I used to run SIP on my phones, but i find i get more out the phone using SIP |
| 16:02.01 | stromboli | *skinny |
| 16:02.17 | stromboli | the name hasnt changed |
| 16:02.29 | beardy | Ah. |
| 16:03.23 | stromboli | it's almost like they redesigned skinny and never put out a new example config file |
| 16:03.31 | beardy | Like what, out of interest? (I only have two 7940s for labbing (and using of course)). |
| 16:04.02 | stromboli | phone screen saver |
| 16:04.06 | beardy | Stay around, hopefully someone will po up that knows sooner or later. |
| 16:04.22 | stromboli | phone to phone intercom |
| 16:04.26 | beardy | Yes.. that's the only thing I could think of myself. |
| 16:05.04 | stromboli | i've been digging through the chan_skinny.c |
| 16:07.28 | stromboli | i know not a lot of people use it (skinny) but i cant be the only person with this problem ;) |
| 16:11.16 | *** join/#asterisk TecR0c (~tecr0c@acetow2.lnk.telstra.net) |
| 16:11.48 | TecR0c | hello, i need to do an update on a centos/trixbox production server |
| 16:11.58 | TecR0c | just wanted to get some guide before doing so |
| 16:12.12 | TecR0c | as i have noticed alot of people have had issues |
| 16:12.14 | WIMPy | ~trixbox |
| 16:12.15 | infobot | well, trixbox is SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 16:12.26 | TecR0c | WIMPy, very true |
| 16:12.38 | TecR0c | but i still need to make sure its half secure. |
| 16:13.03 | WIMPy | Unfortunaletly that's probably the only answer you will get. |
| 16:14.05 | TecR0c | no recommendations on how to update the system |
| 16:14.22 | TecR0c | without affecting the voip service? |
| 16:14.47 | TecR0c | or a fix on our to patch the LFI |
| 16:16.09 | *** join/#asterisk knot (~knotsucke@unaffiliated/devemo) |
| 16:18.50 | p3nguin | stromboli: It seems like most people using SCCP phones on Asterisk prefer to use the third-party chan_sccp-b rather than chan_skinny. |
| 16:21.04 | stromboli | understood |
| 16:21.48 | stromboli | i used to use that and then i switched to chan_skinny in Ast 1.6, just wondering if anyone has run into this issue also |
| 16:21.48 | *** join/#asterisk [cannibalera] (~cannibale@201-41-247-4.fnsce703.dsl.brasiltelecom.net.br) |
| 16:22.05 | *** part/#asterisk [cannibalera] (~cannibale@201-41-247-4.fnsce703.dsl.brasiltelecom.net.br) |
| 16:22.25 | ManxPower | stromboli, if you are digging around the code, try #asterisk-dev |
| 16:22.53 | ManxPower | Trixbox is not closed source. That is the only thing infobot is wrong about |
| 16:24.00 | stromboli | ok |
| 16:24.05 | stromboli | thanks for your help |
| 16:27.37 | ManxPower | stromboli, the problem with Cisco is that they want to sell you a PBX as well as phones. They have no interest in making the advanced features available in their non-SIP firmware. Polycom does not sell PBXs. They want their phones to work with as many systems as they can. Cisco has lost the opensource / openstandards phone market. |
| 16:27.37 | gregd | guys, what does it mean if a handset connected to a line can be rang (using its extension), and if i connect spa3102 in place of this handset... the extension is out of service? what im trying to do is to hookup spa3102 to existing company pbx in place of my handset |
| 16:27.46 | *** join/#asterisk mciml (~Ziki@c-68-44-177-83.hsd1.nj.comcast.net) |
| 16:28.15 | ManxPower | gregd, what handset |
| 16:28.20 | mciml | Hey |
| 16:28.29 | gregd | ManxPower: Siemens optiset E |
| 16:28.36 | ManxPower | gregd, does it run SIP? |
| 16:28.51 | gregd | ManxPower: no, AFAIK a regular analogue line |
| 16:28.56 | gregd | "regular" |
| 16:29.18 | ManxPower | I cannot help you. |
| 16:29.19 | stromboli | i would convert back to SIP if i cant fix... but i dont feel like buying SmartNET to get a license for the SW |
| 16:29.53 | ManxPower | stromboli, welcome to Cisco's licensing policies. When I want an upgrade to my Polycom I download it from the polycom site. 8-) |
| 16:30.05 | stromboli | lol |
| 16:30.05 | gregd | or in otherwords... trying to connect spa3102 to Hicom 100 E |
| 16:30.10 | *** join/#asterisk markit (~marco@88-149-177-66.staticnet.ngi.it) |
| 16:30.19 | ManxPower | gregd, I'm not seeing Asterisk in that setup. |
| 16:30.40 | stromboli | well when i got the phones for free, i try to make the best out of it |
| 16:30.55 | gregd | ManxPower: asterisk is the second step, that i've configured already.. work fine with spa3102.. but now got the problem spa3102<->provider |
| 16:31.15 | ManxPower | stromboli, only if your time is free. |
| 16:31.32 | stromboli | ;) |
| 16:31.47 | Kobaz | mmm, that was a good astricon |
| 16:31.50 | Kobaz | time for breakfast |
| 16:41.48 | *** join/#asterisk [cannibalera] (~cannibale@201-41-247-4.fnsce703.dsl.brasiltelecom.net.br) |
| 16:42.42 | stromboli | lmao i found the problem |
| 16:43.50 | stromboli | it's not clear in the config file, now you have to separate [lines] from [devices] |
| 16:44.07 | stromboli | before [devices] had lines |
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| 17:10.46 | *** join/#asterisk Dovid (Dovid@213.8.121.90) |
| 17:10.49 | Dovid | j #asterisk-il |
| 17:11.27 | Dovid | hello all |
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| 17:42.44 | carrar | HARRO |
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| 18:13.46 | *** join/#asterisk zogg35 (~Ryan@89.211.98.132) |
| 18:16.09 | zogg35 | Is this an appropriate place for a simple support question? |
| 18:16.19 | WIMPy | ~ask |
| 18:16.20 | 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. |
| 18:17.21 | zogg35 | I'm upgrading from 1.4 to 1.8 and I noticed that when I do "core set verbose 10" I don't get nearly the helpful messages I got with 1.4. For example, it doesn't tell me when incoming calls happen, when it plays specific sound files, etc. |
| 18:17.39 | zogg35 | How do I get the old style of messages to help me debug? |
| 18:28.51 | ManxPower | zogg35, did you read the 1.4, 1.6, and 1.8 UPGRADE.txt files? |
| 18:29.10 | zogg35 | Not completely. I'll go check those. Thanks! |
| 18:29.41 | ManxPower | zogg35, Also check /etc/asterisk/logger.conf It may or may help you, but you should look at the 1.8 version of that |
| 18:30.20 | zogg35 | Sounds good, thanks a lot |
| 18:30.20 | ManxPower | zogg35, it is good to check all three versions since many docs on the web and on voip-info are 1.4 or 1.6 specific without saying they are |
| 18:30.31 | zogg35 | I've noticed that. ;) |
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| 18:57.40 | morphieous | I am using asterisk 1.6, how to detect if someone answered your call so you can play a message over pstn line |
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| 19:00.06 | *** part/#asterisk test34 (~test34@unaffiliated/test34) |
| 19:00.38 | morphieous | currently it's making calls and plays the message but if there is a delay then some parts of the message gets lost |
| 19:00.41 | morphieous | any idea |
| 19:04.27 | [TK]D-Fender | morphieous: What kind of line and how are you using it? |
| 19:09.56 | morphieous | [TK]D-Fender:asterisk 16, pstn line, trunk ZAP, created a call file placed in /var/spool/asterisk/outgoing/ |
| 19:10.59 | [TK]D-Fender | morphieous: What device to use the line? |
| 19:11.20 | [TK]D-Fender | morphieous: And there is no such thing as Zaptel for 1.6 |
| 19:12.55 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 19:22.01 | *** join/#asterisk morphieous (add18b7e@gateway/web/freenode/ip.173.209.139.126) |
| 19:23.10 | morphieous | [TK]D-Fender: following I have in my call file |
| 19:23.28 | morphieous | [TK]D-Fender: channel: Local/1234567890@from-internal maxretries: 1 retrytime: 60 waittime: 60 callerid: "testid" <4161234567> context: playmsg extension: s priority: 1 Set: DIAL=1234567890 Set: ID=299 Set: MSG= data: |
| 19:24.05 | [TK]D-Fender | [15:10]<[TK]D-Fender>morphieous: What device to use the line? |
| 19:24.07 | [TK]D-Fender | [15:11]<[TK]D-Fender>morphieous: And there is no such thing as Zaptel for 1.6 |
| 19:24.31 | morphieous | [TK]D-Fender: In my extensions_custom.conf I have the following: [playmsg] exten => s,1,Answer exten => s,2,Wait(2) exten => s,3,Playback(custom/playsilent) exten => s,4,Wait(2) exten => s,5,Playback(custom/msg) exten => h,1,Hangup |
| 19:25.04 | *** join/#asterisk typename (~andy@sprocom09.ee.cooper.edu) |
| 19:26.17 | morphieous | [TK]D-Fender: Genuine Digium TDM04B 4 FXO PCI card for Asterisk |
| 19:27.04 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 19:27.13 | [TK]D-Fender | morphieous: the "default" with Zap/DAHDI FXO's is to consider the call as ANSWERED the moment it is placed. It does not detect any actual "answer". |
| 19:27.45 | [TK]D-Fender | morphieous: The only option for this is to set "callprogress=yes" in their channel configs. This risks random hangups however |
| 19:30.39 | a1fa | in last day I got asterisk scanned from 10 different sources |
| 19:30.40 | a1fa | :) |
| 19:30.45 | a1fa | damn |
| 19:31.19 | lanning | asterisk scanned? or do you mean SIP scanned? |
| 19:31.42 | a1fa | sip* |
| 19:31.46 | [TK]D-Fender | lanning: He stated the target, not the means |
| 19:31.46 | a1fa | lanning: good catch |
| 19:32.00 | a1fa | hehe.. yeah.. bastards |
| 19:32.09 | lanning | I get SIP scanned all the time, and don't even have an asterisk server at this IP... |
| 19:32.23 | [TK]D-Fender | lanning: And who was talking about YOU? :p |
| 19:32.23 | a1fa | they tried to bruteforce extensions again |
| 19:32.29 | lanning | :) |
| 19:32.37 | a1fa | i need to bring it back in-house |
| 19:33.06 | [TK]D-Fender | a1fa: Have you considered securing your server with more than just a "keep out" sign hanging over an open door? :p |
| 19:33.24 | p3nguin | If you want to be bothersome (and if you have the resources to spare), answer the calls to those extensions and do funny things with the calls. |
| 19:33.25 | a1fa | when i get a new computer i will virtualize my firewall and split it into * and firewall |
| 19:33.41 | a1fa | [TK]D-Fenderk: i usually add the spammers into my drop bucket |
| 19:33.47 | a1fa | but otherwise it is wide open |
| 19:33.48 | lanning | my snort report shows a lot of "Sipvicious" scans |
| 19:34.01 | a1fa | because it is a web server too :) |
| 19:34.20 | a1fa | but considering the issues i've been having with different phones, I will just insource it |
| 19:34.47 | morphieous | [TK]D-Fender: what is the best solution for making a call and play the message when some picks up. Any pointers thanks |
| 19:35.06 | a1fa | morphieous: bat phone? |
| 19:35.11 | [TK]D-Fender | morphieous: I just gave you the only answer you have with what you've got |
| 19:41.08 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 20:10.17 | ManxPower | morphieous, see "core show application dial" what you want to do won't work on FXO ports without dialplan work that is beyond your abilities |
| 20:10.53 | [TK]D-Fender | ManxPower: Umm.. no.. CALLPROGRESS |
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| 20:35.44 | vinhdizzo | [TK]D-Fender: u here? i have questions on wav and mp3 support in asterisk |
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| 20:38.13 | *** join/#asterisk geneg1 (~gene@199.243.243.122) |
| 20:46.32 | [TK]D-Fender | ~ask |
| 20:46.32 | 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. |
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| 21:22.08 | naimols | does anyone have asterisk 1.8 setup with google talk via jabber? i keep on getting "res_jabber.c:1693 aji_act_hook: JABBER: encryption failure. possible bad password." but the secret in jabber.conf is correct |
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| 21:37.08 | *** join/#asterisk twanny796 (~twanny@80.77.205.35) |
| 21:37.57 | twanny796 | any mISDN experts? |
| 21:38.07 | WIMPy | ~ask |
| 21:38.07 | 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. |
| 21:38.47 | twanny796 | cannot get my 'ISDN BRI PCI HFC - S' card working |
| 21:39.22 | *** join/#asterisk dandate2 (~gtejkgjke@180.190.196.198) |
| 21:39.44 | dandate2 | i have a moh file 24 minutes long and * refuses to play it. is it the length that is a problem? |
| 21:39.45 | WIMPy | What Kernel? What problem? |
| 21:40.46 | twanny796 | kernel 2.6.32.15-ipfire |
| 21:41.10 | twanny796 | misdn_info: no card found, please load drivers |
| 21:41.46 | WIMPy | And what happens if you try to load the driver? dmesg? |
| 21:42.00 | naimols | does anyone have asterisk 1.8 setup with google talk via jabber? i keep on getting "res_jabber.c:1693 aji_act_hook: JABBER: encryption failure. possible bad password." but the secret in jabber.conf is correct |
| 21:42.23 | twanny796 | WIMPy: I loaded hfcpci, still with the same error |
| 21:42.31 | WIMPy | dandate2: Dont' think so, but there are some things Asterisk doesn't like, like Tags. And converting to 8ksps mono certainly helps. |
| 21:42.58 | WIMPy | twanny796: dmesg |
| 21:45.30 | WIMPy | ~pb |
| 21:45.30 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 21:45.45 | WIMPy | twanny796: Try that next time |
| 21:59.46 | twanny796 | WIMPy: yep, forgot, sry |
| 21:59.48 | twanny796 | http://pastebin.ca/1977587 |
| 22:01.57 | WIMPy | twanny796: I can't find any occasion of hfcpci being loaded in there. |
| 22:04.57 | twanny796 | WIMPy: moment |
| 22:05.10 | *** join/#asterisk Lantizia (~lantizia@erebus.seaquake.net) |
| 22:06.10 | twanny796 | WIMPy: but dmesg is boot messages, I'm loading mISDN by hand right now |
| 22:06.25 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-123-250.lsanca.fios.verizon.net) |
| 22:06.41 | twanny796 | WIMPy: I can paste lsmod? |
| 22:06.55 | WIMPy | Then look at dmesg after you tried loading hfcpci. |
| 22:08.54 | vinhdizzo | are there specific requirements for wav, gsm, and mp3 files to be played in asterisk's Background() command? do i need additional packages installed to play them? |
| 22:09.41 | twanny796 | I reloaded hfcpci, but still I cannot see it in dmesg, but as far as I know dmesg gives you the boot up messages : http://unixhelp.ed.ac.uk/CGI/man-cgi?dmesg+8 |
| 22:10.21 | WIMPy | dmesg is the kernel log. |
| 22:10.44 | vinhdizzo | I get silence when playing wav, and this is the debug log http://pastebin.com/h3TcH8Jq |
| 22:11.46 | WIMPy | vinhdizzo: Never put an extension of a sample file in your dialplan. Asterisk will look for the most apropriate match. |
| 22:12.11 | vinhdizzo | WIMPy: ok. i get similar results without specifying sextensions |
| 22:12.20 | vinhdizzo | will post debug in a sec |
| 22:12.39 | twanny796 | WIMPy: how can I force the irq and io to hfcpci? |
| 22:13.42 | WIMPy | twanny796: You can't. With PCI it's always detected. |
| 22:16.23 | twanny796 | WIMPy: I'll paste lspci, moment |
| 22:16.32 | vinhdizzo | without .wav extensions http://pastebin.com/hDQ39s8e |
| 22:16.52 | vinhdizzo | state it needs to be in mono2 |
| 22:17.53 | WIMPy | vinhdizzo: Then mix it down to one channel. And unless you need more, downsaple to 8000s/s as well/ |
| 22:18.23 | vinhdizzo | WIMPy: I did that before, and still didn't work. Will re-do now just to make sure again |
| 22:19.00 | twanny796 | http://pastebin.ca/1977598 |
| 22:19.12 | twanny796 | WIMPy: ^^^ |
| 22:20.40 | WIMPy | twanny796: Still waiting to see what happens if you load the driver. |
| 22:23.50 | twanny796 | WIMPy: http://pastebin.ca/1977600 |
| 22:25.32 | WIMPy | twanny796: What do you think that line means? |
| 22:25.43 | WIMPy | hfcpci: Unknown parameter `irq' |
| 22:29.09 | twanny796 | WIMPy: that's when I tried to pass the irq parameter |
| 22:29.43 | WIMPy | Such parameter dosn't exist and prevents the module from loading at all. |
| 22:30.49 | twanny796 | WIMPy: looks like there's no hfcpci loaded anywhere??? |
| 22:31.36 | WIMPy | I haven't seen any evidence of it, But just told you why it wouldn't load. |
| 22:35.26 | vinhdizzo | WIMPy: thanks, playback is resolved. removing .wav extensions and getting to mono and 8000 Hz works |
| 22:35.58 | vinhdizzo | for mp3 playback on 1.8, I suppose the above requirements are also needed? |
| 22:36.53 | WIMPy | vinhdizzo: It's certainly beneficial, but I dont think it is required. |
| 22:37.15 | vinhdizzo | So the requirements only apply to wav and not mp3s? |
| 22:37.52 | WIMPy | Sorry, but I can't give you a definitive answer. |
| 22:38.06 | vinhdizzo | ok |
| 22:38.14 | twanny796 | WIMPy: but hfcpci is a driver not a kernel module, yes? ;) |
| 22:38.48 | WIMPy | twanny796: What's the difference for you? |
| 22:39.07 | WIMPy | To me it's both. |
| 22:39.22 | twanny796 | WIMPy: but I can see hfcpci in lsmod |
| 22:39.57 | WIMPy | ok, then it should appear somewhere in your dmesg as well. |
| 22:46.07 | twanny796 | WIMPy: do you know the command to list the parameters that can sent to a module? |
| 22:46.43 | WIMPy | modinfo <modulename> |
| 22:50.03 | twanny796 | WIMPy: I'm already passing debug=0x9, but no errors are found in messages, could it be the card is faulty? |
| 22:50.59 | WIMPy | There is no message what so ever when you load the module? |
| 22:52.54 | WIMPy | Ok, just tried it. It does indeed not log anything if it finds no hardware. |
| 22:53.18 | WIMPy | So I guess the PCI ID of your card may be unknown to it. |
| 22:53.19 | vinhdizzo | Does asterisk recognize Background(~/sound/mysound) or Background($HOME/sound/mysound)? |
| 22:53.38 | vinhdizzo | Or do absolute paths need to be specified? |
| 22:54.08 | twanny796 | WIMPy: can I force it, somehow? |
| 22:54.09 | WIMPy | twanny796: Try lspci -v to find the cards ID. |
| 22:55.18 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 22:57.59 | twanny796 | WIMPy: http://pastebin.ca/1977622 |
| 23:01.25 | WIMPy | Err, isn't it about device 00:0a.0? The Tiger thing? |
| 23:04.01 | twanny796 | WIMPy: yes, but this is 00:0b.0 |
| 23:05.29 | WIMPy | Now I don't know what that 1ae7:0520 thing is, but the Tiger3XX seems to be e159:0001 which is not listed in hfcpci. |
| 23:06.29 | WIMPy | So I guess the only thing you can do is try to see if it's compatible with one of the supported cards by copying it's definition in the driver or replace the card. |
| 23:11.51 | twanny796 | WIMPy: the Tiger3XX has the wcfxo driver |
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| 23:12.17 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 23:15.21 | WIMPy | So that is not ISDN as listed? |
| 23:16.29 | twanny796 | WIMPy: I don't think so, I use it as FXO channel |
| 23:17.03 | WIMPy | Well, if it's the unknown one, that seems to be VERY unknown. |
| 23:17.34 | WIMPy | You could still try out if itBut the comment still applies. |
| 23:17.41 | WIMPy | ^^ So I guess the only thing you can do is try to see if it's compatible with one of the supported cards by copying it's definition in the driver or replace the card. |
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| 23:19.15 | twanny796 | WIMPy: this I cannot understand "...by copying it's definition in the driver or replace the card.". Where? |
| 23:22.48 | WIMPy | In the source. |
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| 23:55.11 | WIMPy | Oh, while we're on misdn... |
| 23:56.40 | WIMPy | leifmadsen: I took a peek at the new book yesterday. IIRC chan_misdn is listed as 'usable' there. Are yu sure it is? If I understand it correctely it will only work with misdn1 which has been dead |
| 23:56.45 | WIMPy | for 2 or 3 years now. |
| 23:56.57 | leifmadsen | shrugs |
| 23:57.00 | leifmadsen | I didn't write that part |
| 23:57.10 | WIMPy | ok |
| 23:57.44 | WIMPy | I can't even select cahn_misdn on my test machine, which is running misdn2. |
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