| 00:16.28 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.23) |
| 00:18.07 | Katty | evening |
| 00:18.25 | Katty | file: happy birthday! |
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| 00:44.05 | *** join/#asterisk raden (~Jon@24-240-51-238.dhcp.stpt.wi.charter.com) |
| 00:44.27 | raden | anyone recommend a SMS sip phone ? |
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| 00:48.10 | s14ck | file: congratz! |
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| 00:51.41 | [TK]D-Fender | raden: You seem to be combining terms that should not be combined. |
| 00:51.54 | [TK]D-Fender | SMS is a PSTN/cell concept. SIP is SIP |
| 00:52.00 | raden | [TK]D-Fender, ? |
| 00:52.12 | [TK]D-Fender | SIP has SIP messaging. |
| 00:52.20 | [TK]D-Fender | There is no such thing as "SMS" in SIP |
| 00:52.24 | Penguin | In other words, there is no POTS VoIP phone. |
| 00:53.20 | [TK]D-Fender | Penguin: How do I adjust the wings on my bicycle to get FM radio? |
| 00:53.33 | Penguin | Did you read the documentation? |
| 00:53.53 | [TK]D-Fender | I don't have a big IQ! |
| 00:54.07 | [TK]D-Fender | How do I adjust the wings on my bicycle to get FM radio? |
| 00:54.23 | [TK]D-Fender | Penguin: The circle is complete! |
| 00:54.34 | raden | why are all these voip providers pushing SMS |
| 00:55.06 | Penguin | The purpose of an ITSP is to link us VoIP users to the PTSN. |
| 00:55.18 | Penguin | That kind of, by definition, means they use the PSTN. |
| 00:55.31 | [TK]D-Fender | raden: then forward SIP MESSAGING as "SMS" |
| 00:56.03 | [TK]D-Fender | radewDo not use the term SIP SMS. there is no such thing. The facct your provider takes those SIP messages and forwards them on in such a manner is besides the point |
| 00:57.04 | [TK]D-Fender | raden: So if you want SIP messaging.. go look for devices that support it |
| 00:57.36 | [TK]D-Fender | raden: Several series of hard-phones do, as do many soft-phones |
| 00:58.04 | raden | [TK]D-Fender, interstiing |
| 01:02.34 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.23) |
| 01:10.48 | jeev | fender, what's the best way to figure out the cause of these iax issues regarding it going unreachable/reachable quite often? watch poke/pong ? |
| 01:12.58 | jeev | i've had to build multiple paths to eachother, it used to be stable but now it is only stable over the VPN connection, i've even put both on a public IP, literally no bandwidth, ping is around 25ms.. still dropping. |
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| 01:25.02 | jeev | it's not doing it now, i'll leave it alone. |
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| 01:27.26 | [TK]D-Fender | [21:10]jeevfender, what's the best way to figure out the cause of these iax issues regarding it going unreachable/reachable quite often? watch poke/pong ? <- the cause? Qualify not MAKING it. that is the ONLY cause, because that is what those statu's REPORT |
| 01:28.39 | jeev | fender, yea.. i understand qualify isn't making it, there is nothing wrong in the connection that i can see. there are no training errors, no dropped packets, nothing. i even put it on a pbulic ip |
| 01:28.50 | jeev | since i put them on a public ip, it did it once, in the beginning, it's been 30 minutes, i'll wait and see. |
| 01:33.03 | [TK]D-Fender | things aren't making it on time. |
| 01:33.10 | [TK]D-Fender | lost or late. |
| 01:33.13 | [TK]D-Fender | Take your pick. |
| 01:33.15 | [TK]D-Fender | Both bad |
| 01:33.59 | [TK]D-Fender | Of course if you're dealing with fixed IP's you shouldn't normally have to care about this at all. |
| 01:34.12 | [TK]D-Fender | And are simply doing this to yourself. |
| 01:35.43 | jeev | why? just work with IP and no auth ? |
| 01:36.17 | *** join/#asterisk jasonwert (~w3rt@71-89-135-86.static.aldl.mi.charter.com) |
| 01:37.47 | redotis | I think google only allows so many simultaneous inbound/outbound calls. How can you check to see if a line/channel is unavailable and, if so, use another device for outbound calls? |
| 01:38.17 | [TK]D-Fender | jeev: you have a QUALIFY timeout issue. That has nothing to do with auth |
| 01:38.23 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.73) |
| 01:38.34 | jeev | so dont qualify? i dont understan your fixed ip comment |
| 01:38.36 | [TK]D-Fender | jeev: You don't seem to have grasped its purpose or its consequence |
| 01:39.18 | [TK]D-Fender | jeev: Qualify A; can keep a NAT forwarding open for a remote client behind NAT without fixed forwarding. |
| 01:39.46 | jeev | ah |
| 01:39.47 | [TK]D-Fender | jeev: and B: Track the response time and KILL your ability to dial the peer |
| 01:40.15 | [TK]D-Fender | jeev: You don't NEED "A" |
| 01:40.24 | [TK]D-Fender | jeev: and B is kicking you in the nuts. |
| 01:40.28 | jeev | i see |
| 01:40.35 | [TK]D-Fender | jeev: So ask yourself why you did this to yourself |
| 01:40.37 | jeev | it wasn't happening before though but ok i see |
| 01:40.43 | jeev | fender, obviously because i don't know better. |
| 01:41.04 | [TK]D-Fender | jeev: You really need to look at what these things do for you before you turn them on... |
| 01:41.13 | [TK]D-Fender | jeev: You've been using * a very long time now... |
| 01:41.58 | [TK]D-Fender | jeev: So if you have an actual problem and no qualify, what is the consequence? |
| 01:43.01 | jeev | if i have a network issue without qualify enabled/ |
| 01:44.05 | [TK]D-Fender | jeev: What part of my question was unclear? |
| 01:44.39 | jeev | if i have an actual problem without qualify ENABLED ? it will go unreachable |
| 01:44.49 | *** join/#asterisk danjenkins (~danjenkin@63.133.202.2) |
| 01:44.50 | [TK]D-Fender | no |
| 01:45.13 | [TK]D-Fender | "unreachable" is a status reserved for things being tracked by qualify |
| 01:45.29 | [TK]D-Fender | no qualify = UNMONITORED |
| 01:45.59 | jeev | ah |
| 01:46.31 | [TK]D-Fender | So, the peer ISN'T checked for anything. What happens? |
| 01:47.24 | jeev | it goes unmonitored. |
| 01:48.03 | Penguin | "goes" indicates that it was one thing and then becomes another. When qualify is disabled, it's just simply unmonitored. |
| 01:48.39 | jeev | well, going from qualify to not having qualify goes unmonitored. from monitored to unmonitored. |
| 01:48.47 | jeev | i guess i stupidly use qualify to determine if there are any network issues |
| 01:49.28 | [TK]D-Fender | Says who? |
| 01:49.48 | [TK]D-Fender | first the question of how big until it is a "problem" should be configured |
| 01:49.53 | [TK]D-Fender | and then the actual consequence. |
| 01:50.08 | [TK]D-Fender | You don't seem to understand what the impact actually is |
| 01:50.27 | [TK]D-Fender | A "status" means nothing on it's own. It has to have a CONSEQUNCE attached to it |
| 01:50.35 | Penguin | its own |
| 01:50.38 | [TK]D-Fender | So ..... what happens? |
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| 02:09.16 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.73) |
| 02:35.02 | phix | ok what are the hardware requirements for Asterisk? My setup will consist of 12 internal SIP handsets, 4 concurrent external SIP connections, [ua]law codecs. |
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| 03:11.13 | redotis | phix, you could run that on a pentium IV with 256mb of ram. |
| 03:11.54 | redotis | I think google only allows so many simultaneous inbound/outbound calls. How can you check to see if a line/channel is unavailable and, if so, use another device for outbound calls? |
| 03:12.47 | *** join/#asterisk mae_tae (7960eecf@gateway/web/freenode/ip.121.96.238.207) |
| 03:21.16 | phix | you would get an unavailable error code when attempting to make the call, if that occurs just use another trunk |
| 03:24.49 | phix | Use a dialing macro, after the dial command add something like: exten => s,n,Goto(s-${DIALSTATUS},1) |
| 03:24.52 | phix | exten => s-CHANUNAVAIL,1,Macro(dialout,SIP/AnotherTrunk,${ARG1}) |
| 03:25.44 | phix | Although you should read up what return code it gives, as CHANUNAVAIL by also be returned if the number you dialed is engaged, look it up :) |
| 03:26.25 | phix | Alternativally if you know how many concurrent calls google supports then set a limit in your sip config file |
| 03:26.39 | phix | exten => s,n,Goto(s-${DIALSTATUS},1) |
| 03:26.44 | phix | call-limit=4 or something |
| 03:26.57 | phix | heh copy paste fail |
| 03:28.45 | phix | You get the idea :) |
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| 03:42.16 | Penguin | Ooooooooooh... you told him to go look up something on his own. You're going to hell for that. |
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| 04:55.32 | redotis | Hey...thanks a lot phix. I will look all that up. :) |
| 04:55.36 | redotis | I really appreciate it. |
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| 04:59.46 | MrBar | hi |
| 05:02.07 | MrBar | someone know why global_reg_timeout not work for sip_reg_timeout in scheduler? |
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| 07:46.24 | *** join/#asterisk TazzNZ (~pieter@nz.insync.za.net) |
| 07:46.46 | TazzNZ | hey Everyone - quick question - can someone please explain to me why SIP has a registration |
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| 07:47.38 | TazzNZ | the only reason I can see is for state checking - "Is my peer availble ?" |
| 07:47.55 | TazzNZ | other than that, I can't figure it out |
| 07:50.36 | TazzNZ | oh - and grats on moving back Qwell :) |
| 08:00.28 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 08:11.03 | ChannelZ | Registration is so a peer can tell asterisk (or whatever other system) what its IP is so that system knows how to reach it. |
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| 08:26.33 | fling | ChannelZ: is not 0.1% packet loss to bad for sip? |
| 08:29.55 | *** join/#asterisk vlad_starkov (~vlad_star@195.239.220.131) |
| 08:29.59 | ChannelZ | Well pretty much any packet loss is not great |
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| 08:37.25 | fling | :[ |
| 08:40.27 | ChannelZ | .1% doesn't sound like much, where have you come up with that number? |
| 08:41.31 | *** join/#asterisk Geek-Linux (~mubbashir@mbl-109-36-224.dsl.net.pk) |
| 08:42.28 | Geek-Linux | hi every one, i want to configure live channels audio streams in asterisk. can any one guide me ? |
| 08:42.49 | Geek-Linux | live channels means tv channels e.g bbc |
| 08:45.44 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.244.113) |
| 08:49.19 | TazzNZ | ChannelZ: pretty much what I thought - so it will never be used for call placements ? |
| 08:49.40 | TazzNZ | aka - "call via the registration" |
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| 08:51.20 | ChannelZ | TazzNZ: I don't really understand what you're asking I guess. |
| 08:52.04 | TazzNZ | ChannelZ: let me rephase - is there a way to make a call via asterisk without getting the unathorized back from a peer |
| 08:52.37 | TazzNZ | the 401 response that is sent back |
| 08:52.56 | ChannelZ | Short of not requiring authorization, no |
| 08:53.08 | ChannelZ | That's kind of the point. |
| 08:53.43 | TazzNZ | yeah - I see - I have a feeling my provider is going to disagree with "you" |
| 08:54.01 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 08:54.34 | TazzNZ | but - if you "look at the protocol" - it makes sense (try, then auth) |
| 08:55.37 | ChannelZ | ITSPs typically aren't setup to auth for calls they send to YOU |
| 08:56.10 | TazzNZ | the issue was cause in the other direction |
| 08:56.27 | TazzNZ | Asterisk triggered the DDoS of the ITSP |
| 08:57.04 | ChannelZ | How so |
| 08:57.19 | TazzNZ | not 100% sure - I will get details in about 9 hours |
| 08:57.25 | TazzNZ | night time here now :) |
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| 08:59.41 | TazzNZ | thanks for confirming anyways - I have read the RFC and came to the same outcome |
| 08:59.53 | TazzNZ | but it pays to make 100% sure :) |
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| 09:24.37 | Neoti | hi all |
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| 09:54.33 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 09:54.37 | v0lZy | hi |
| 09:55.11 | v0lZy | I have a question regarding asterisk registering to the same provider twice. |
| 09:55.36 | v0lZy | The story goes a bit like this: |
| 09:56.58 | v0lZy | The VoIP provider bills SIP trunks based on IP. I have 2 public IP's, and I need asterisk to reach the VoIP provider through these two different IP's based on which registration the call comes in/out through |
| 09:58.02 | v0lZy | My idea is to do outbound 1:1 NAT in a way where asterisk for one sip trunk would register to some local IP on the router that gets 1:1 translated to one of the public IP's and routed on that gateway |
| 09:59.40 | v0lZy | my question however pertains to the sip.conf file, more directly to the [general] context where I have the ability to input my public IP... can this parameter that controls which public IP is reported also be put into each context where i configure the connection to the providers SIP trunk? |
| 10:00.25 | v0lZy | I'm guessing that asterisk when behind NAT needs the public IP to be reported by this parameter so that it can send what IP it expects to receive media on. |
| 10:00.48 | v0lZy | but in this case, it needs to report two different IP's |
| 10:01.16 | v0lZy | Is this doable with 1 asterisk instance or should I start looking into a chrooted environment or something? |
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| 10:11.44 | v0lZy | I guess what I am asking is does the 'externip' setting in sip.conf apply to any single provider section of sip.conf or only the [general] section? |
| 10:12.57 | kaldemar | v0lZy: no |
| 10:13.17 | kaldemar | v0lZy: only in [general] |
| 10:21.33 | *** join/#asterisk HasanAtizaz (~hasan@119.155.9.99) |
| 10:23.25 | v0lZy | kaldemar: So there is no way of actually forcing asterisk to use more than 1 public IP address? |
| 10:24.41 | v0lZy | kaldemar: My problem lies in the fact that I want to avoid doing CDR because its more legitimate if the tennant gets it from the phone company. My idea was to use a separate SIP trunk, but they can only create it if i use a different outgoing IP |
| 10:24.46 | kaldemar | not from asterisk. |
| 10:26.42 | HasanAtizaz | if anyone can help me setting up a basic sip |
| 10:27.01 | kaldemar | ~book |
| 10:27.01 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 10:27.06 | kaldemar | HasanAtizaz: ^ |
| 10:27.26 | HasanAtizaz | thank you i tried but call is not going. |
| 10:27.31 | HasanAtizaz | my client is ekiga. |
| 10:28.29 | v0lZy | well, what I was hoping to do was |
| 10:30.14 | v0lZy | kaldemar: What I was hoping to do was to use 1:1 NAT and have asterisk register to my router's IP (router would output a virtual IP on the LAN interface then do 1:1 NAT to a virtual IP on the WAN interface) |
| 10:31.27 | v0lZy | kaldemar: That takes care of the VoIP provider seeing a different IP, but the packets my asterisk will be sending will prboably specify that its listening for media on whatever the externip is set to, correct? |
| 10:32.04 | kaldemar | yes, you would need something outside asterisk. |
| 10:32.17 | v0lZy | kaldemar: but what does that mean for rtp media? |
| 10:32.36 | v0lZy | kaldemar: I mean... can asterisk onyl ever advertise a single ip to which others should send their media? |
| 10:33.52 | v0lZy | kaldemar: does that in my case mean that outgoign conenction and sip registration will be through a different IP, but asterisk will advertise my primary public IP and thats where the media will come in on? If that is the case, then i gather i dont need to open any ports other than those for SIP on my virtual LAN and virtual WAN IP's, because all rtp will pass through primary WAN anyway? |
| 10:34.07 | v0lZy | (am i making any sense btw? :D) |
| 10:34.08 | kaldemar | v0lZy: asterisk uses a single ip. |
| 10:34.39 | kaldemar | you'd need to mangle the packets in that scenario. |
| 10:34.51 | v0lZy | kaldemar: what do you eman by mangle? |
| 10:54.13 | *** join/#asterisk jploh (~textual@112.204.31.96) |
| 10:57.23 | v0lZy | Allow me to draw this |
| 10:58.44 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 11:13.29 | v0lZy | kaldemar: sorry about my drawing skills and terminology, but this is what i had in mind: http://snag.gy/mXCGE.jpg |
| 11:14.36 | kaldemar | you were just wasting time with that. |
| 11:15.05 | v0lZy | :( |
| 11:15.43 | v0lZy | What i wanted to know was, is it wrong to expect that NAT in this situation will be the solution, because its not a proxy |
| 11:15.55 | v0lZy | i could also bridge it i suppose |
| 11:16.20 | v0lZy | but if its a bridge, then i cant send packets to it i suppose |
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| 11:38.23 | mirela666 | hello, how to set a pattern match for 3 or 4 digits (not moreor not less)? thanks |
| 11:39.19 | kaldemar | mirela666: _XXX and _XXXX |
| 11:39.44 | mirela666 | so I need to have everything doubled |
| 11:40.27 | kaldemar | well, not everything. you can of course put a goto in the first priority of those extensions and use a common extension. |
| 11:44.04 | mirela666 | kaldemar: so I match _XXX and goto _XXXX,1, would be best solution? |
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| 12:20.35 | Geek-Linux | can any one tell me what this means : "config.c:263 find_my_host: Unable to find host configuration for host 'msnrstatus'." |
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| 12:45.24 | BorjaGVO | Hi all! Can anybody tell me what a semicolon means when putting it at the end of a GotoIf (in the priorirty position...in this case "1:2"): exten => s,1,GotoIf($["${OUT_${DIAL_TRUNK}:4:4}" = "A2B/"]?custom-freepbx-a2billing,${OUTNUM},1:2) |
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| 12:46.28 | kaldemar | BorjaGVO: core show application GotoIf |
| 12:48.51 | BorjaGVO | kaldemar: it doesn't say anything about this syntax on the priority section |
| 12:48.53 | BorjaGVO | :-/ |
| 12:49.02 | BorjaGVO | At least I cannot see it. |
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| 12:50.42 | wdoekes | : <-- colon, not semicolon |
| 12:50.44 | wdoekes | and |
| 12:50.45 | wdoekes | $ asterisk -nrx 'core show application GotoIf' | grep : |
| 12:50.45 | wdoekes | GotoIf(condition?[labeliftrue][:labeliffalse]) |
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| 12:52.05 | DataWraith | i cannot make a call outside, with asterisk version 11.5, but the sip-trunk is registered at the provider. |
| 12:52.53 | BorjaGVO | wdoekes: thanks for the correction. Yes, that applies for the label/extension section. But does it refer to the priority section aswell (after the last comma)? |
| 12:54.25 | BorjaGVO | wdoekes: oh |
| 12:54.26 | BorjaGVO | sorry |
| 12:55.09 | BorjaGVO | now I interpreted it as it should be. Number two refers to priority number two inside that same extension |
| 12:55.14 | BorjaGVO | Thanks |
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| 13:22.16 | Katty | morning. |
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| 13:22.35 | malcolmd | aloha |
| 13:22.53 | coppice | oh no it isn't |
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| 13:50.40 | BorjaGVO | Here http://goo.gl/No2doJ I see AGI "GET VARIABLE". However, in the following code it takes a second argument. What does it mean? I can't find any documentation about it. -- if ($dynamic_idconfig = intval($agi->get_variable("IDCONF", true))) { -- |
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| 13:52.09 | [TK]D-Fender | BorjaGVO: that is not an AGI parameter. |
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| 15:27.59 | bananapie | I am running latest asterisk cetified 1.8. I added 't38pt_udptl = yes,fec,maxdatagram=400' to my sip.conf, but when I call from my ATA to my SIP provider crossing asterisk, the re-INVITE arrives from my ATA with t38 in SDP and leaves the other side of asterisk with only pcmu. The re-invite is sent after asterisk bridges the local channel out of the call. Any ideas? |
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| 15:51.00 | bananapie | The refusal is because of AST_T38_REFUSED |
| 15:51.04 | bananapie | but I am not sure why. |
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| 16:02.29 | qakhan | hi all. i setup support line for 3 exts. if someone call on support line all 3 exten ring. if no one pickup the call then call goes to exten 1 voice mail. is there any way a call can go all 3 voice mail as it ring on all 3 exten |
| 16:03.55 | kaldemar | qakhan: what does "core show application VoiceMail" tell you? |
| 16:04.27 | [TK]D-Fender | qakhan: it's your dialplan.. it goes wherever you told it to do. |
| 16:04.31 | [TK]D-Fender | g0* |
| 16:06.05 | qakhan | kaldemar so i can do VoiceMail(1001@vm&1002@vm&1003@vm) |
| 16:06.11 | qakhan | right? |
| 16:07.07 | [TK]D-Fender | qakhan: If those are valid box #'s in valid VM contexts |
| 16:07.10 | kaldemar | assuming those are valid mailboxes, yes. |
| 16:07.57 | qakhan | thanks :) |
| 16:09.37 | pyromaniac | hi. I am experimenting with the chan_dongle module. I have two different huawei devices, one E1550 (shows up as E173) and one E160. Both work in mobile partner, but on asterisk I only get one-way audio. |
| 16:09.43 | pyromaniac | is there something I am missing? |
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| 16:22.54 | qakhan | VoiceMail(1001@vm&1002@vm&1003@vm) is not send in VM to all exten it sends only to 1st exten(3223) |
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| 16:27.38 | *** join/#asterisk przerull (~philip@philzerull.com) |
| 16:30.39 | przerull | so my dialplan keeps hanging up when it tries to load the grammar file. How do I debug this? google wasn't particularly helpful. https://gist.github.com/przerull/6870821 |
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| 16:41.01 | *** join/#asterisk kaspar (~kaspar@p50811E05.dip0.t-ipconnect.de) |
| 16:41.47 | kaspar | Hello! I have a problem to place an outgoing sip call. Registration works fine. The SIP call is rejected with 403 Unauthorized. |
| 16:41.59 | kaspar | I have a trace of a working call with some unknown SIP phone. |
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| 16:42.24 | kaspar | Main difference is that the phone sends REGISTER in the INVITE options field, while asterisk does not. |
| 16:42.39 | kaspar | Does anyone have an idea on how to enable that in asterisk? |
| 16:43.34 | przerull | did you register asterisk to the terminating sip server? |
| 16:43.56 | przerull | (i.e. a register => line in sip.conf) |
| 16:44.10 | kaspar | yep, registration is OK |
| 16:45.16 | [TK]D-Fender | you can't have a registration inside an invite. They are 2 completely different actions |
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| 16:46.40 | przerull | kaspar: did you use the sip settings your carrier provided? Typically they will provide you with the sip.conf settings |
| 16:47.02 | kaspar | [TK]D-Fender: hm, this SIP phone has REGISTER in it's Allow: header |
| 16:47.20 | przerull | also, you are dialing SIP/yourcarrier/phonenumber right? |
| 16:47.33 | [TK]D-Fender | kaspar: that sounds like an OPTIONS listing. |
| 16:47.47 | przerull | yeah that's not uncommon that's just the server saying that it accepts register requests |
| 16:48.15 | kaspar | przerull: actually I'm working for the carrier, so they don't provide any sip settings yet (other than server, username, pwd) |
| 16:49.06 | kaspar | [TK]D-Fender: it's in the INVITE message header... looks weird to me too |
| 16:49.21 | kaspar | przerull: I'll double-check dialing |
| 16:49.21 | [TK]D-Fender | kaspar: I wouldn't concentrate on that being the important factor |
| 16:49.26 | kaspar | ok |
| 16:49.36 | [TK]D-Fender | kaspar: Go prove the rest of your peer is sane, that you are indeed registered, and then show us all of it |
| 16:49.53 | kaspar | will do |
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| 16:52.32 | przerull | dir |
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| 16:58.06 | *** join/#asterisk dzu (~user@ppp-46-244-243-97.dynamic.mnet-online.de) |
| 16:58.22 | dzu | Hi there! |
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| 16:59.37 | dzu | I think I need a little help as I'm trying to get our old asterisk 1.4.21 installation working again on Ubuntu 12.04 |
| 17:00.19 | dzu | I use this card: 05:01.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) |
| 17:00.47 | dzu | Unfortunately I did not do the original configuration and currently I'm somewhat lost in converting the old zaptel based configuration to dahdi |
| 17:01.13 | [TK]D-Fender | dzu: No version below 1.8 series is supported at this point. |
| 17:01.48 | dzu | yes, I'm switching to 1.8.10.1~dfsg-1ubuntu1 now |
| 17:02.05 | dzu | as I was saying, dahdi_hardware sees the card: |
| 17:02.19 | dzu | # dahdi_hardware |
| 17:02.19 | dzu | pci:0000:05:01.0 wcb4xxp+ 1397:08b4 Junghanns QuadBRI ISDN card |
| 17:02.19 | dzu | <PROTECTED> |
| 17:02.20 | *** join/#asterisk danjenkins (~danjenkin@63.133.202.2) |
| 17:02.42 | dzu | from within asterisk I see this dahdi status: |
| 17:02.53 | dzu | taurus*CLI> dahdi show status |
| 17:02.53 | dzu | Description Alarms IRQ bpviol CRC Fra Codi Options LBO |
| 17:02.53 | dzu | DAHDI_DUMMY/1 (source: HRtimer) 1 UNCONFI 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) |
| 17:02.53 | dzu | B4XXP (PCI) Card 0 Span 1 UNCONFI 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) |
| 17:02.56 | dzu | B4XXP (PCI) Card 0 Span 2 UNCONFI 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) |
| 17:02.59 | dzu | B4XXP (PCI) Card 0 Span 3 UNCONFI 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) |
| 17:03.04 | dzu | B4XXP (PCI) Card 0 Span 4 UNCONFI 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) |
| 17:03.07 | dzu | taurus*CLI> |
| 17:03.11 | dzu | does this look feasible? |
| 17:03.11 | Penguin | Stop. |
| 17:03.15 | Penguin | Use a pastebin. |
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| 17:03.37 | dzu | sorry, I thought those few lines would still be ok, but I'll respect this of course |
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| 17:04.32 | dzu | Here it is: http://pastebin.com/PzQptHqY |
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| 17:07.15 | [TK]D-Fender | dzu: Seems available but unconfigured |
| 17:07.45 | dzu | yep, I also see this unconfigured - but how do I "configure" it? |
| 17:08.16 | [TK]D-Fender | dzu: edit your dahdi configs |
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| 17:10.14 | dzu | [TK]D-Fender: ok, that's exactly where I'm struggling - I only have the zaptel configs from the old installation |
| 17:10.43 | dzu | [TK]D-Fender: so obviously I'm missing something here that worked in the old config. |
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| 17:11.57 | kaspar | [TK]D-Fender, przerull: Dial string was wrong... I used Dial(SIP/${EXTEN}@peername) which didn't work. |
| 17:12.23 | dzu | [TK]D-Fender: I admit openly that I miss the understanding of what I need to look for. I'd need a "if this x is missing, then you need to change y" type of instruction |
| 17:12.46 | dzu | [TK]D-Fender: I googled a lot but could not find a step-by-step procedure to get this done, so this is why I ask |
| 17:13.17 | [TK]D-Fender | dzu: You need chan_dahdi configured to use the channel. for the zaptel config names, those are dead. |
| 17:15.11 | Penguin | Isn't the config file's format like exactly the same? |
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| 17:26.49 | qakhan | [TK]D-Fender and kaldemar when i leave message in voice mail after that i get message |
| 17:27.08 | qakhan | <PROTECTED> |
| 17:27.28 | qakhan | 1001 has VM but 1002 does has VM |
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| 18:10.45 | dzu | [TK]D-Fender: Thanks for the pointer - I waded through the settings and got it to work now |
| 18:11.14 | dzu | [TK]D-Fender: I needed to disables SRTP on my snom with the new 1.8 asterisk but this also works again. |
| 18:11.53 | *** join/#asterisk navaismo (~navaismo@189.191.249.45) |
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| 18:34.49 | bananapie | Hello I am calling the a sip address in the following style sip/user@foreigndomain.tld, given that I am directly calling the URI there is no corresponding peer in sip.conf, and I think this is preventing asterisk from allowing t38 passthrough on the channel. Does this make sense? How can I tell asterisk to allow t.38 on this channel? |
| 18:40.22 | *** join/#asterisk mokmeister (~mokmeiste@86-40-248-153-dynamic.b-ras2.lmk.limerick.eircom.net) |
| 18:41.45 | Katty | jigs through |
| 18:42.23 | HasanAtizaz | SIP/2.0 401 Unauthorized |
| 18:42.47 | HasanAtizaz | SIP/2.0 501 Method Not Implemented |
| 18:42.47 | HasanAtizaz | Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKd8b3c9e0-ed2d-e311-9da9-0019d19f2950;received=192.168.1.20;rport=5060 |
| 18:42.48 | HasanAtizaz | From: <sip:hasan@192.168.1.20>;tag=46a0c9e0-ed2d-e311-9da9-0019d19f2950 |
| 18:42.48 | HasanAtizaz | To: <sip:hasan@192.168.1.20>;tag=as3d147e47 |
| 18:42.48 | HasanAtizaz | Call-ID: 2e5a740a-be2d-e311-9da9-0019d19f2950@debian |
| 18:42.48 | HasanAtizaz | CSeq: 1243 PUBLISH |
| 18:42.49 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:42.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:42.50 | HasanAtizaz | Server: Asterisk PBX 1.6.2.9-2+squeeze11 |
| 18:42.52 | HasanAtizaz | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO |
| 18:42.54 | HasanAtizaz | Supported: replaces, timer |
| 18:42.56 | HasanAtizaz | Content-Length: 0 |
| 18:43.22 | HasanAtizaz | any idea what could be the problem. |
| 18:44.47 | Qwell | ~upgrade asterisk |
| 18:44.47 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 18:46.30 | Penguin | bananapie: If the only problem is that there is no defined peer entry, create a peer entry. |
| 18:49.06 | bananapie | Penguin, I just tested for the sake of testing, and it does the same thing. |
| 18:50.14 | bananapie | I have a SIP Trunk connected to my Asterisk box ( 1.8 cert3 ) connected to my SPA122 ( which I reach through a remote sip proxy ). When I call, the SPA122 requests t.38 and asterisk refuses with 488 Not acceptable here. It never tries to negociate t.38 on the sip provider leg of the call. |
| 18:50.37 | bananapie | when I call from my cellphone via the sip provider, via asterisk, via the proxy to the SPA122* |
| 18:51.20 | bananapie | When the SPA122 makes an outbound call, it negociates t.38 without issue. But when it receives the call, asterisk doesn't seem to want to negociate t.38 with the device. |
| 18:51.45 | bananapie | I have t38pt_udptl=yes in [general] in sip.conf. |
| 18:52.29 | bananapie | Also, when I receive a call from my sip provider and answer it with app_fax, t.38 is negociated ok. So the problem seems to be when my asterisk calls a device which requests t.38, it refuses the request. |
| 18:53.20 | bananapie | I have eliminated all the local channels to dial directly, there are no more 'dial local/ext@context' in the call. |
| 18:54.41 | bananapie | So for some reason, my asterisk won't accept a t.38 re-invite from a callee when it bridges two SIP channels. |
| 18:56.36 | bananapie | it doesn't matter if I call sip/user@outsidedomain.tld or sip/user@localsippeer, either way it gives 488 not acceptable here. |
| 18:57.23 | Penguin | SIP/peer/extension |
| 18:57.40 | Penguin | (not that it will fix your issue, but that's the correct way to do that.) |
| 18:58.44 | bananapie | Penguin, you are right, it did not fix my issue. I used the syntax indicated and I get the same result. |
| 18:59.10 | bananapie | I fixed my dialplan though for the sake of proper syntax. |
| 18:59.22 | Penguin | I knew that would be the case, but it's still a good idea to use the right syntax to avoid other problems later. |
| 18:59.30 | bananapie | Yes, you are right. |
| 19:00.01 | bananapie | I learned that the hard way with cfengine3. In ver 3, they didn't care about misplaced commas, but in version 3.5 they did. I spent 4 hours updating all my configs :( |
| 19:00.56 | DataWraith | what should be the default syntax for a dialplan which can dial every number to pstn? |
| 19:01.06 | bananapie | I checked if my issue was a timing issue, the final ACK is sent by asterisk to my SIP provider before asterisk receives the SIP RE-INVITE. There is a 5 second delay, so it isn't a race condition. |
| 19:01.43 | Penguin | datawraith: Is your asterisk connected to the PSTN? |
| 19:02.13 | DataWraith | Penguin: i have configured the SIP trunk with the provider data and it could register to it. but i cannot dial outside |
| 19:02.32 | DataWraith | i use asterisk 11.5 with FreePBX |
| 19:02.50 | Penguin | Since you've erroneously used the term "SIP trunk," that tells me that you do not have your asterisk connected to the PSTN. |
| 19:02.56 | Penguin | ~freepbx |
| 19:02.56 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:04.58 | [TK]D-Fender | [15:00]DataWraithwhat should be the default syntax for a dialplan which can dial every number to pstn? <- "every number" is dangerously vague. You should know what kinds of patterns are sane to actually pass to the PSTN and make an OUTBOUND ROUTE to match them |
| 19:05.59 | *** join/#asterisk navaismo (~navai_000@189.191.249.45) |
| 19:06.02 | bananapie | it gets better. Asterisk sends a re-invite to the SIP provider asking for G711, it's like asterisk thinks it can be a t.38 gateway even though it can't... |
| 19:06.15 | bananapie | is t.38 affected by disallow=all and allow=g711? |
| 19:06.23 | DataWraith | thx, infobot |
| 19:07.02 | DataWraith | thx, fender |
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| 19:29.26 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:30.48 | bananapie | When two SIP calls are bridged and one of the two calls sends a re-invite to asterisk, does the core handle it or is it handled entirely within chan_sip? |
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| 19:59.30 | redotis | Anyone have any idea what this is about when receiving google voice calls with asterisk? No application 'SendDTMF' for extension |
| 20:00.17 | *** join/#asterisk danjenkins (~danjenkin@63.133.202.2) |
| 20:00.39 | navaismo | seems like you dont have the senddtmf application loaded |
| 20:01.43 | bananapie | module load app_senddtmf.so ? |
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| 20:03.38 | redotis | thanks guys |
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| 20:40.01 | eirirs | can one use SCCM phones at asterisk? :p |
| 20:42.28 | Penguin | Never heard of it. |
| 20:49.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:04.02 | eirirs | sorry, I meant SCCP |
| 21:04.23 | WIMPy | yes |
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| 21:10.00 | Penguin | eirirs: I use chan-sccp-b. It works pretty alright. |
| 21:10.43 | Penguin | I recently upgraded to 4.1-STABLE, and haven't had time to find any bugs yet. |
| 21:11.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:11.43 | Penguin | eirirs: Which asterisk version are you going to be using with SCCP phones? |
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| 21:16.54 | eirirs | ver 11 |
| 21:16.58 | redotis | Does google voice let you adjust the callerid name and or number for outgoing? If so, is it a setting in voice.google.com? |
| 21:17.02 | eirirs | got some cisco sip phones I dunno about transforming to SCCP |
| 21:17.02 | redotis | thanks guys |
| 21:18.50 | Penguin | eirirs: I don't know if there were any improvements to the digium chan_skinny in 11, but in 1.8 and earlier it sucked a banana. |
| 21:19.22 | Penguin | eirirs: If chan_skinny is still bad in 11, chan-sccp-b 4.1 will work in asterisk 11. |
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| 21:23.39 | Penguin | Anyone happen to know if digium's chan_skinny received any improvements after 1.8? |
| 21:26.00 | WIMPy | It did. |
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| 21:37.34 | Penguin | As I was going to say, before I was so rudely interrupted... |
| 21:37.58 | Penguin | The last time I tried chan_skinny, it barely sent and received calls. It certainly didn't do anything else. |
| 21:40.29 | paulc | Speaking of Cisco phones.. what's the general concensus on the 79xx series? I played with one YEARS ago.. it was "alright".. but haven't used them in a long time.. latest bigger install used all SPA-508Gs and they're not bad.. but I recall the 79xx being a pig to configure. Any thoughts to the contrary? |
| 21:41.29 | [TK]D-Fender | paulc: Still pigs. |
| 21:43.03 | paulc | Thought as much.. I'm sure I've got one in a box in my garage somewhere.. the SPA-508Gs aren't bad - we did a bunch of scripting+database stuff behind the scenes to automate provisioning and configuration, works really well.. Happy enough having one on my desk (but my home setup is better than at work - go figure!) |
| 21:44.07 | [TK]D-Fender | SPA'a are OK... a good value away from North America. There Polycom is still the strongest quality choice. |
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| 22:12.09 | paulc | Polycom <3 - from my old IP650 to the new VVXs.. can't go wrong with Polycom.. (although XML provisioning isn't my favourite thing on the block) |
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| 22:26.28 | TheGuy_ | if I have a queue setup that members login to with a *3 key press and I enter them in the queues.conf with queue member priorities, are they automatically logged into that queue? |
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