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| 01:41.26 | freetown2 | hi all, I suddenly started having a problem with sip peers from my provider |
| 01:41.39 | freetown2 | they would all become unreachable. |
| 01:42.57 | freetown2 | tcpdump shows the provider sending packets back with "SIP/2.0 407 Proxy Authentication Required" but asterisk sip debug shows nothing but constant attempts retransmitting REGISTER |
| 01:43.26 | freetown2 | sip show channels shows a growing list of active sip dialogs |
| 01:43.38 | freetown2 | any clues as to what is going on? |
| 01:44.40 | WIMPy | Have you tried turning it off and on again? |
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| 01:46.13 | freetown2 | WIMPy, you mean restarting asterisk? many times... |
| 01:46.56 | WIMPy | And it stays deaf? |
| 01:48.30 | freetown2 | all sip peers are unreachable. I tried upgrading to asterisk 1.6 by upgrading the entire hardy distro to lucid and I got one slight improvement...the first number becomes reachable but the rest no and it appears that the growing list of channels then makes the first sip peers unreachable too. |
| 01:49.12 | WIMPy | 1.6 is like stone age. |
| 01:50.05 | freetown2 | okay...I could try upgrading to precise and whatever asterisk they have...but is it normal to have a growing list of sip channels? I am now up to 135 active sip dialogs |
| 01:50.19 | freetown2 | 140 and counting |
| 01:50.29 | freetown2 | multiple dialogs for each sip peer |
| 01:50.33 | WIMPy | No. There's something going terribly wrong. |
| 01:51.12 | freetown2 | i never had this before and there has been no configuration change...but could this be configuration related? |
| 01:52.13 | WIMPy | I did have that isse a long time ago with phones publishing their state. |
| 01:52.28 | WIMPy | For your version that would be in the far future. |
| 01:53.05 | freetown2 | no phones connecting to this...it serves as a go between for a cisco pbx and the provider |
| 01:53.34 | jameswf | Go go power hackers |
| 01:55.49 | jameswf | Solid possibility you are in the process of.... How do the kids say it these days.... Getting owned |
| 01:56.21 | freetown2 | rooted? |
| 01:56.55 | jameswf | Nothing that cool... Rootkits are so 1998. |
| 01:57.26 | jameswf | Close your firewall see if it all drops |
| 01:59.34 | freetown2 | jameswf, as in stop filtering? |
| 02:00.15 | jameswf | As in FILTER ALL THE THINGS |
| 02:00.24 | jameswf | Drop all traffic |
| 02:00.49 | jameswf | See of all traffic is coming from a single ip,/subnet |
| 02:00.59 | freetown2 | i have strict rules as to who can connect to asterisk. Except for the provider, nothing is allowed to connect. tcpdump does not show any other connections for that matter |
| 02:03.32 | freetown2 | all the sip dialogs are to the provider btw... |
| 02:04.29 | freetown2 | i forgot to say that I am getting registration timeouts for all sip peers from the provider |
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| 03:10.39 | wrxed | Looking for some assistance with H323. Specifically trying to place calls from my Asterisk server to an Altigen box that is configured to use H323. I can get the phones to ring / setup, but I'm not getting audio. Using OOH323 installed from the addons menu. Anyone have any common settings that I may have missed? |
| 03:11.08 | wrxed | my experience with H323 is limited to the last 4 hours of googling... so i'm sure I'm missing something, i Just don't know what. |
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| 03:16.35 | tulga | I see extension name in queue_log, I need extension number like LOCAL/100 in queue_log. where configure it? |
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| 04:53.24 | gavimobile | im very concerned about my new cisco router. my previoous router was a worse situation but im suspiciius my router is causing minor dissconections. http://pastebin.com/K4SMSEHE |
| 04:53.34 | gavimobile | my trunks go and cmoe |
| 04:53.36 | gavimobile | come* |
| 04:54.28 | gavimobile | is thhis normal? or is it normal to have a solid connection with my trunks without any lagging or disconnectiosn? |
| 04:54.33 | gavimobile | dissconnectiosn* |
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| 05:12.07 | gavimobile | my dns settings may have been off.. |
| 05:12.10 | gavimobile | back to more testing |
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| 08:38.32 | linocisco | is Grandstream UCM610x and cisco 7945G compatiable |
| 08:38.35 | linocisco | ? |
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| 08:59.55 | linocisco | is Grandstream UCM610x and cisco 7945G compatiable |
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| 09:51.15 | giorgiodinapoli | imagine a sip client and a pbx are talking to each other but no voice can be heard. how can we find out why, is there a good step by step plan? |
| 09:56.37 | eirirs | NAT |
| 09:56.47 | eirirs | involved? |
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| 11:06.27 | dan_j | Hi. If i can see sip packets on the asterisk server, using 'tcpdump port 5060', any idea why i can't see them when using 'sip set debug ip IPADDRESS'? |
| 11:06.38 | dan_j | Its like its being filtered somewhere, but iptables is totally open. |
| 11:07.30 | XATRIX | Hi guys, i know it's not a support channel for Elastix. But can you advice me ? I'm trying to get my PBX talk my local lang. Which one archive should i donwload from http://downloads.asterisk.org/pub/telephony/sounds/ ? |
| 11:07.38 | XATRIX | alaw,ulaw, gsm ? |
| 11:08.16 | XATRIX | And what is it for ? I thought there should be one independent codec format |
| 11:08.54 | dan_j | Download all and let asterisk sort things out. |
| 11:10.10 | XATRIX | How ? I tried to put all the files to /var/lib/asterisk/sounds/it |
| 11:10.26 | XATRIX | But the asterisk still talks to me english |
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| 11:10.51 | XATRIX | Moreover i did echo "language=it" >> /etc/asterisk/sip_general_custom.conf |
| 11:22.08 | ayrjola | XATRIX: dont konw much about elastix, but n free pbx used to be languagecode under general section on webpage |
| 11:24.25 | XATRIX | Yes, it's there, but i don't need to have multilang setup. Only IT voice |
| 11:26.23 | dan_j | Is it possible to store the hangup reason for all sip calls? |
| 11:27.05 | dan_j | XATRIX: have you tried copying the IT files into the default location, and overriding all the english files? |
| 11:27.29 | XATRIX | Yes, but nothing changed |
| 11:27.48 | XATRIX | If i remove EN directory - then i have no voices at all |
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| 11:27.58 | XATRIX | But if i override em, nothing changes O_O |
| 11:28.27 | dan_j | So move EN directory. Download IT directly, and rename it EN |
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| 11:31.46 | Egyptian[Home] | http://fpaste.org/122234/40680614/ <-- can someone enlighten me with the issue i need to fix? |
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| 11:48.03 | lnb | fresh install of ubuntu 12.04 x64, ran install_prereq install, screen is full of lines like open: 200631; closed: 540315; defer: 8; conflict: 8 |
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| 12:11.57 | Zogot | ahoyhoy |
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| 12:15.15 | Zogot | on the asterisk 12 branch, the mysql schemas are missing |
| 12:15.18 | Zogot | they have been moved? |
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| 12:15.28 | Zogot | http://svn.asterisk.org/svn/asterisk/branches/12/contrib/realtime/mysql/ http://svn.asterisk.org/svn/asterisk/branches/11/contrib/realtime/mysql/ |
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| 12:29.21 | puzzled | hi |
| 12:29.26 | XATRIX | Guys, if i terminate my call with "Put caller on hold forever" , how can i get this call back after i finish to talk to the previous caller ? |
| 12:30.29 | XATRIX | Or, call termination - does total disconnect from the call queue ? |
| 12:34.15 | dan_j | Is there any way to reload the iax bindport without a full restart? when i do iax2 reload, i get this 'Ignoring bindaddr on reload' |
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| 13:23.24 | wrxed | Looking for some assistance with H323. Specifically trying to place calls from my Asterisk server to an Altigen box that is configured to use H323. I can get the phones to ring / setup, but I'm not getting audio. Using OOH323 installed from the addons menu. Anyone have any common settings that I may have missed? |
| 13:23.52 | wrxed | my experience with H323 is limited to the about 5 hours of googling... so i'm sure I'm missing something, i Just don't know what. |
| 13:24.39 | WIMPy | The problem is exactely the same as with SIP. |
| 13:25.03 | WIMPy | Just that the counterpart to reinvites should be safe. |
| 13:26.20 | wrxed | could you elaborate on that a bit please? |
| 13:27.01 | WIMPy | It usually doesn't work when NAT is involved. |
| 13:27.30 | WIMPy | SIP has various workarounds. I Doubt there has been much done in that respect in th H323 channel. |
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| 13:34.38 | wrxed | Splendid. I was afraid of that |
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| 14:12.03 | Kobaz | does the security event stuff exist in 1.8? |
| 14:14.41 | [TK]D-Fender | * 1- IIRC |
| 14:14.43 | [TK]D-Fender | 10 |
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| 14:16.13 | mjordan | [TK]D-Fender: you recall correctly |
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| 14:20.48 | Egyptian[Home] | router :) |
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| 14:40.10 | Kobaz | mm |
| 14:40.22 | Kobaz | just wondering what to do with fail2ban stops working |
| 14:40.28 | Kobaz | i guess an upgrade to 12 is in order at some point |
| 14:41.00 | Kobaz | s/stops/when it stops/ |
| 14:41.12 | puzzled | hi. does anyone know where I can find a list of NANPA area codes that are non-US? |
| 14:42.30 | [TK]D-Fender | http://www.nanpa.com/pdf/NANP_Member_Country_Maps.pdf |
| 14:42.47 | Kobaz | nifty |
| 14:43.18 | [TK]D-Fender | It's like they have a SITE or something! |
| 14:43.34 | puzzled | [TK]D-Fender: doh, thanks |
| 14:44.54 | puzzled | weird, Canada is missing from that list. Guess we'll see an invasion soon then :) |
| 14:45.08 | [TK]D-Fender | puzzled: Go further back on thier site |
| 14:45.20 | [TK]D-Fender | puzzled: can & US have dedicated pages |
| 14:45.29 | puzzled | cool, thanks |
| 14:45.55 | Katty | hi fenderbender. |
| 14:51.40 | *** join/#asterisk sip (~DID@voip.pribroker.com) |
| 14:51.43 | [TK]D-Fender | Katty: Mew |
| 14:51.54 | Katty | [TK]D-Fender: how're you dear |
| 14:52.05 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 14:53.00 | [TK]D-Fender | Katty: Meh. connection issues here driving me nuts... |
| 14:53.51 | lnb | setup odbc.ini/odbcinst.ini with valid credentials, isql -v asterisk-connecter fails with access denied for root@localhost. root is not in odbc.ini |
| 14:54.12 | lnb | why is it trying to connect as root? |
| 14:54.23 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-neucdwhsztfbnriz) |
| 14:57.33 | [TK]D-Fender | Where do we see it saying anything else? |
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| 15:19.16 | lnb | strange, dial our phone number with cell, it rings on cell but nothing on sip phone. Tried changing destination on inbound route, and nothing |
| 15:19.18 | lnb | wtf |
| 15:20.38 | lnb | see the inbound call in cli, but that's it |
| 15:24.27 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:2f70:21b:63ff:fe31:8426) |
| 15:27.09 | lnb | http://pastebin.com/uiMiwu88 |
| 15:35.15 | lnb | i dont get it. Tried sending to other extensions and call is same.. rings on cell, shows up once in cli and then ringing stops and nothing in cli like hangup |
| 15:37.50 | lnb | deleted inbound route for the DID and call still does same thiing |
| 15:37.52 | lnb | thing |
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| 15:43.13 | defswork | SIP peers with status LAGGED - how to clear that ? I think it's ok now but * isnt using it |
| 15:43.23 | defswork | sip trunk |
| 15:44.55 | lnb | this is really off. from itsp route call to a2billing server then -> freepbx server and the call works. Route call direct from itsp -> freepbx server call doesn't come in |
| 15:45.03 | lnb | wtf is causing this? |
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| 15:48.10 | newtonr | defswork, sip reload maybe? I'm not sure |
| 15:48.20 | defswork | yeah I did that and it cleared |
| 15:48.24 | newtonr | cool |
| 15:48.28 | defswork | but is there a way it will self clear ? |
| 15:50.10 | *** join/#asterisk wrxed (~rrittgarn@MERCHANTS-E.edge2.Miami1.Level3.net) |
| 15:50.43 | mjordan | If a SIP peer is lagged, it is because the time to qualify the peer is exceeding what you configured for your qualify setting. |
| 15:51.18 | mjordan | Each time Asterisk qualifies the peer, it will recalculate the RTT of the OPTIONS request versus the max qualify time for that peer. |
| 15:51.37 | mjordan | It will only magically fix itself if whatever is causing the lag goes away |
| 15:51.57 | defswork | hmm - ok - it looked like the latency issue was resolved hours ago though |
| 15:51.59 | mjordan | Generally, if you are getting LAGGED peers, you either (a) have a setting that is too small for the peer, or (b) you have network problems. |
| 15:52.17 | defswork | there was some routing issues with our provider |
| 15:52.58 | defswork | * was showing a constant 2351ms - never varied - so it looked like it wasn't re-reporting it |
| 15:53.50 | newtonr | did you look to see if the OPTIONS requests were still being sent out and responded to? |
| 15:54.11 | defswork | no - sorry - something to remember |
| 15:54.28 | defswork | I never got down to sip debugging it |
| 15:54.56 | newtonr | yeah, and if you are getting responses, but the response time is stuck, make sure to capture debug logs and pcap then report it on the tracker. |
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| 16:37.10 | Quastor | Hi, doe somebody knows if it's possible to keep the Original CID when transferring a call to an other extension? |
| 16:39.58 | [TK]D-Fender | do a blind transfer and not an attended transfer |
| 16:41.37 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 16:45.40 | Quastor | Yes ok I have to test that but you also know that most people use an attended, but when they do the transfer you are connected with the Original caller and not with the extension who did the transfer |
| 16:46.29 | Quastor | the same problem with call pickup |
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| 16:56.51 | Kobaz | who'se good with crisco routers |
| 16:57.17 | Kobaz | i got's a nat question.... i have like 40 phones behind a cisco C881G-V |
| 16:57.44 | Kobaz | they are registering to a remote host, the host gets the reg, sends back options, and then the cisco is not passing back the options response, so the phones are unreachable |
| 17:03.37 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:06.24 | Kobaz | oh yay i fixed it |
| 17:06.27 | Kobaz | turned off nat for sip |
| 17:17.22 | Kobaz | for future reference: no ip nat service sip udp port 5060 |
| 17:17.28 | Kobaz | will fix sip on cisco |
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| 18:20.53 | infinity1 | who are some good pay as you go sip trunk providers? |
| 18:21.52 | [TK]D-Fender | vitelity, voip.ms, les.net |
| 18:22.29 | *** part/#asterisk Eugene (eugene@kashpureff.org) |
| 18:23.09 | infinity1 | [TK]D-Fender: thanks for the tip! |
| 18:24.04 | infinity1 | [TK]D-Fender: seems the party-line list has changed. been out of the buesiness for ages. glad i asked :) |
| 18:24.28 | Qwell | infobot is a bit out of date, I suspect... |
| 18:24.28 | [TK]D-Fender | A few others have certainly fallen in the public view |
| 18:24.33 | [TK]D-Fender | Some right off the map |
| 18:24.38 | Qwell | ~itsplist-us |
| 18:24.38 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com, http://www.voicepulse.com/connect/, http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
| 18:24.44 | Qwell | Disclaimer: I do not agree with the above. |
| 18:24.59 | infinity1 | i would remove teliax. you can't sign up on the website anymore. |
| 18:25.11 | [TK]D-Fender | Voicepulse has had enough periods with issues taht I hear very little about them anymore... |
| 18:25.17 | [TK]D-Fender | Them too |
| 18:25.53 | Qwell | ~itsplist-us |
| 18:25.54 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
| 18:26.04 | Qwell | updated, but somebody that doesn't have a stake in one of the companies should update the list. |
| 18:26.21 | infinity1 | Qwell: which do you represent ? :) |
| 18:26.42 | Qwell | infinity1: Not so much represent, but my employer owns/runs sipstation.com |
| 18:28.14 | infinity1 | Qwell: well, if you're here backing the community, they must be good right? :) |
| 18:31.59 | Qwell | infinity1: I'm here because I'm an addict; there is no PBX Developers Anonymous support group. |
| 18:46.45 | Katty | paces to |
| 18:47.05 | Katty | paces fro |
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| 18:58.51 | mjordan | Qwell: I'm fine with anyone updating ye olde infobot. If it recommends silly things, I think whoever updated it will get yelled at collectively :-) |
| 18:59.22 | Qwell | mjordan: well, I updated it to remove old/bad listings. I just don't feel comfortable making any changes myself, given the obvious bias. |
| 18:59.50 | Katty | i'll make changes! |
| 19:00.12 | Katty | me and infobot have a long happy history together, of doing silly things. |
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| 19:04.15 | ChannelZ | infobot: be serious is <reply> no. |
| 19:04.15 | infobot | okay, ChannelZ |
| 19:04.24 | Katty | infobot: be serious |
| 19:04.24 | infobot | no. |
| 19:04.30 | Katty | infobot: why not? |
| 19:04.30 | infobot | why not? |
| 19:04.33 | Katty | infobot: yes. |
| 19:04.34 | infobot | somebody said yes was the opposite of no |
| 19:04.41 | Katty | infobot: you're hopeless. |
| 19:04.54 | Katty | infobot: DON"T YOU WALK AWAY FROM ME AND NOT ANSWER. |
| 19:05.08 | Katty | infobot: i love you |
| 19:05.08 | infobot | You love you? |
| 19:05.13 | Katty | sobs |
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| 20:44.36 | *** join/#asterisk mutable_malachi (~mmalachi@unaffiliated/malachiconstant/x-837457) |
| 20:45.10 | mutable_malachi | Hi guys and girls and other. :-) |
| 20:47.39 | mutable_malachi | I'm doing a bit of grunt work for an apartment building with 100 analog phone lines coming into a server room. New construction. The server room hasn't been built yet. |
| 20:47.53 | TazzNZ | poor you |
| 20:47.58 | mutable_malachi | TazzNZ: :-) |
| 20:48.07 | TazzNZ | why not VoIP ? |
| 20:48.11 | Qwell | Why analog? |
| 20:48.17 | TazzNZ | and you serve them internet over the same line |
| 20:48.27 | mutable_malachi | Not my decision. Retirement home. My preference would have been SIP everything. |
| 20:48.36 | TazzNZ | that explains a lot : |
| 20:48.39 | mutable_malachi | Yeah. |
| 20:48.39 | TazzNZ | :) |
| 20:48.42 | WIMPy | A new building with analogue lines??? |
| 20:48.55 | mutable_malachi | WIMPy: Analog to the server room. |
| 20:49.04 | Chainsaw | Grandma needs to be able to use http://www.gporetro.com/img/products/gpo-746-button_03_b.jpg you know. |
| 20:49.08 | mutable_malachi | Where I'll employ asterisk to connect the calls via SIP. |
| 20:49.12 | WIMPy | Bad enough |
| 20:49.18 | mutable_malachi | I agree. |
| 20:49.27 | mutable_malachi | But it was the unsatisfactory compromise that we came to. |
| 20:49.36 | Qwell | There was a *worse* solution? |
| 20:49.46 | mutable_malachi | Yeah. Hard lines through the telco. |
| 20:49.51 | mutable_malachi | Retirement home foots the bill. |
| 20:50.02 | WIMPy | How would that be worse? |
| 20:50.05 | TazzNZ | you need 4 of these http://www.digium.com/en/products/telephony-cards/analog/24-port |
| 20:50.16 | TazzNZ | 5 - sorry |
| 20:50.16 | Qwell | TazzNZ: 5 |
| 20:50.33 | TazzNZ | yeah - "early" morning here :) |
| 20:50.40 | Chainsaw | TazzNZ: Coffee will make it better. |
| 20:50.49 | TazzNZ | Working on that Chainsaw :) |
| 20:50.54 | mutable_malachi | <3 coffee |
| 20:50.58 | mutable_malachi | TazzNZ: Thank you! |
| 20:51.43 | TazzNZ | do you even get a card that can handle more than 24 ? |
| 20:51.48 | TazzNZ | I don't think so |
| 20:51.52 | mutable_malachi | Not to my knowledge. |
| 20:52.13 | WIMPy | channel banks. |
| 20:52.45 | mutable_malachi | WIMPy: Like those Adtran things? |
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| 20:53.05 | TazzNZ | but they are $$$, from memory |
| 20:53.34 | mutable_malachi | How much do the digium cards cost? |
| 20:55.25 | TazzNZ | about 1800 each |
| 20:55.27 | TazzNZ | us |
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| 20:56.08 | Qwell | Why would you not get PRIs though? |
| 20:56.21 | WIMPy | Don't know if adtran do them. I'm was referring to the ones that do T1<>Analogue, not SIP GWs. |
| 20:56.22 | mutable_malachi | Do you mean through the telco? |
| 20:56.26 | TazzNZ | the phones are to the rooms |
| 20:56.31 | TazzNZ | not to the ITSP |
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| 20:56.32 | Qwell | mutable_malachi: instead of analog, yes. |
| 20:56.46 | mutable_malachi | We are, on the ITSP side. |
| 20:57.01 | TazzNZ | I would make that SIP tbh |
| 20:57.25 | mutable_malachi | Well, we have a couple options. The ITSP offers PRI and SIP and we haven't priced it out yet. |
| 20:57.29 | Qwell | wait, so it's analog to the rooms? |
| 20:57.32 | mutable_malachi | Yeah. The rooms are analog and that's probably not changing because old folks home. |
| 20:58.53 | mutable_malachi | I'm not exactly thrilled about this given that it's new construction but whatever. |
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| 21:10.57 | marsmars | Hi @all, I have some probs with an asterisk 11 on a router. For example I have softphone and call it the softphone from an ISDN phone via chan_capi. The IAX2 packages are going to the softphone and the softphone answers. (RX and TX) This could I see via dtrace funtions of the AVM Fritz!Box, the modified Router. The asterisk server does not show this RX Packages at the CLI with hight verbose, only the TX. Yesterday I have been here and [ |
| 21:12.34 | mutable_malachi | It looks like adtran channel banks would be somewhat cheaper than the Digium cards. |
| 21:13.09 | WIMPy | You still need a card. |
| 21:13.57 | mutable_malachi | Right, for FXO? |
| 21:14.14 | WIMPy | No for the channelbank. |
| 21:14.38 | mutable_malachi | Right, to connect the channel bank to the asterisk boxes? |
| 21:14.42 | WIMPy | But you could use GWs instead. Might limit your possibilities, tho. |
| 21:14.52 | WIMPy | yes |
| 21:17.32 | mutable_malachi | WIMPy: And do I understand correctly that what it needs is an FXO port? |
| 21:18.02 | WIMPy | An? As in single? |
| 21:18.21 | WIMPy | And does that still refer to the cahnnel bank? |
| 21:19.00 | mutable_malachi | Single, yeah, on the asterisk box side. |
| 21:19.14 | WIMPy | You need one T1 port per channel bank. |
| 21:19.45 | mutable_malachi | Oh, stellar. I'm trying to keep this under $12000 because it's what shorttel quoted us for the proprietary crap. |
| 21:20.15 | mutable_malachi | But that doesn't sound bad. |
| 21:20.30 | WIMPy | PRI cards don't cost much. |
| 21:20.35 | mutable_malachi | nods. |
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| 21:42.49 | mutable_malachi | So okay, I think I understand this correctly. I need channel banks (probably 5 24-ports) with FXS ports, one port for each line, and then 5 T1 ports? |
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| 21:47.28 | WIMPy | yes |
| 21:47.52 | mutable_malachi | WIMPy: Thank you! :-) |
| 21:54.44 | mutable_malachi | http://spidermux.com/ |
| 21:54.48 | mutable_malachi | What do you guys think of these things? |
| 21:55.40 | WIMPy | Do you get drivers for them? |
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| 22:12.39 | mutable_malachi | WIMPy: Looking into that now. |
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