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| 00:47.00 | carrar | moo |
| 00:48.51 | ChannelZ | clucks |
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| 00:56.19 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 03:48.17 | twitchnln | evening everyone |
| 03:51.23 | twitchnln | anyone ever setup cisco with both fxo and fxs as stations and trunk? |
| 03:53.28 | twitchnln | i've gotten the fxs registered and playing nice, but when a comes in via pstn it's coming into from-sip-external instead of the specified trunk context |
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| 04:05.26 | twitchnln | anyone got a clue as to how to get the fxo dial peer to register? |
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| 04:18.49 | p3nguin | twitchnln: You never even bothered to say what you're working with. |
| 04:19.33 | twitchnln | sorry, asterisk 1.4, cisco 1760/4fxd-did/2fxo |
| 04:19.57 | twitchnln | and a typing impediment.. apparently |
| 04:24.28 | twitchnln | relevant configs located @ http://pastebin.ca/2135871 |
| 04:27.41 | p3nguin | You're missing a name for your peer in sip.conf. |
| 04:28.19 | twitchnln | sorry, not missing, was cut-n-paste from freepbx... |
| 04:28.32 | twitchnln | not direct from sip.conf |
| 04:29.20 | p3nguin | http://pastebin.com/BxMqkvbH |
| 04:29.24 | p3nguin | We don't support FreePBX here. |
| 04:29.40 | twitchnln | i know |
| 04:29.52 | p3nguin | Now what was the problem again? |
| 04:30.27 | p3nguin | Your call isn't matching the peer? |
| 04:30.57 | twitchnln | inbound from fxo is coming into incorrect context |
| 04:31.08 | p3nguin | So it probably isn't matching the peer. |
| 04:31.29 | p3nguin | Pastebin a sip debug of a call which experiences the problem. |
| 04:41.47 | twitchnln | http://pastebin.ca/2135876 |
| 04:43.12 | twitchnln | it's matching the dial-peer but not the credentials |
| 04:43.34 | twitchnln | thus i believe asterisk is rejecting call |
| 05:01.26 | [TK]D-Fender | Found no matching peer or user for '192.168.42.2:62614' |
| 05:01.35 | [TK]D-Fender | It is NOT matching a peer |
| 05:03.35 | twitchnln | when you said peer, i was thinking on the dial-peer side, sorry my confusion, so how do i get it to match the trunk i created? |
| 05:05.23 | p3nguin | There is no "trunk." It's a peer. And it isn't matching. |
| 05:05.45 | p3nguin | I don't even know what a "dial-peer" is. |
| 05:06.29 | p3nguin | Did you use the peer entry I put in the pastebin? |
| 05:08.36 | twitchnln | no, please shoot me a link |
| 05:08.46 | p3nguin | (2329.15) <p3nguin> http://pastebin.com/BxMqkvbH |
| 05:10.11 | *** join/#asterisk kayfox (~kayfox@xheotris.zerda.net) |
| 05:11.01 | twitchnln | dropping the username out doesn't resolve issue, |
| 05:11.36 | p3nguin | What you showed me was incomplete. What I gave you was complete. |
| 05:12.28 | p3nguin | I guess you could always change to a user instead of a peer and match on username instead of IP/port. |
| 05:13.29 | twitchnln | i know initially i got into a hurry and dropped the [cisco_fxo], in your updated settings you also changed nat to no and dropped the username, what is difference between no & never on nat? I know that 7940/60s require the never |
| 05:13.36 | twitchnln | but not why |
| 05:15.06 | p3nguin | no = Use NAT mode only according to RFC3581 |
| 05:15.15 | p3nguin | never = Never attempt NAT mode or RFC3581 support |
| 05:15.29 | twitchnln | cool |
| 05:16.07 | p3nguin | In more recent versions, the definition of no was updated to "Use rport if the remote side says to use it." |
| 05:17.14 | p3nguin | Since you are on the LAN, it probably doesn't make a bit of difference. |
| 05:17.24 | twitchnln | right on |
| 05:17.44 | p3nguin | But changing from peer to user would make a difference. |
| 05:18.15 | p3nguin | Peer matches IP/port; user matches username and only allows calls inbound to asterisk. |
| 05:19.22 | p3nguin | If you run 'sip show peers' do you even see the peer cisco_fxo? |
| 05:21.39 | twitchnln | not after changing to user |
| 05:21.57 | twitchnln | which has same results, no matching peer found |
| 05:22.32 | p3nguin | When you change to a user, it is no longer a peer, so show peers has no reason to list it. sip show users would, though. |
| 05:23.23 | twitchnln | it's there |
| 05:28.29 | kaldemar | a peer also matches to username. |
| 05:29.06 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 05:35.27 | twitchnln | switched back to peer, added insecure=port,invite and it's routing correctly |
| 05:35.50 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:36.32 | twitchnln | thanks for your time p3nguin, and the new nat knowledge |
| 05:40.18 | p3nguin | Since when does a peer match username? It has been IP/port since the beginning of time. |
| 05:42.30 | ChannelZ | the sample config either tells the truth or lies. |
| 05:44.31 | p3nguin | Do you see something in the sample other than peer matching IP address and user matching username? |
| 05:45.37 | ChannelZ | yes |
| 05:46.03 | ChannelZ | http://pastebin.com/MRCMa4U7 |
| 05:46.33 | p3nguin | That's what I see in mine, too, and it says what I said. |
| 05:46.34 | ChannelZ | Or, #2 is just unclear |
| 05:46.46 | ChannelZ | (it's also got a typeo) |
| 05:46.59 | p3nguin | I noticed that, too. |
| 05:47.42 | ChannelZ | I guess what it's supposed to mean "checks the From: address" is the hostname/ip given? |
| 05:48.03 | p3nguin | type=peer has always been IP/port. |
| 05:48.30 | ChannelZ | Yes but I'm saying on the surface the explanation is contradictory |
| 05:48.58 | p3nguin | I agree that it is somewhat unclear. |
| 05:49.14 | ChannelZ | So I'm asking what did they really mean to say in #2? |
| 05:49.22 | p3nguin | "From: address" literally means the address in the From: |
| 05:51.00 | p3nguin | What confuses me is how #2 and #3 are separated. |
| 05:51.10 | kaldemar | p3nguin: since always. |
| 05:51.15 | p3nguin | Does that indicate a two step process? |
| 05:51.31 | *** join/#asterisk SewerCow (~sewercow@i121-117-252-75.s05.a008.ap.plala.or.jp) |
| 05:51.54 | ChannelZ | I remember looking at the code once to try and decypher the explanation but don't remember now |
| 05:52.26 | p3nguin | It seems to me that 2 and 3 are kind of the same thing. |
| 05:56.17 | ChannelZ | Well but one mentions the From and the other the INVITE specifically. No matter which way you cut it the whole thing is very poorly defined. I think The Book even says it differently and is even more vague |
| 05:57.20 | p3nguin | I think Olle defined it best, but I don't have his post handy right now. |
| 05:59.03 | p3nguin | Right now what we do is match on IP/port for peers and From: -- Olle E. Johansson |
| 05:59.07 | ChannelZ | Actually I guess it's more definitive in the New Book: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html#DeviceConfig_id291081 |
| 05:59.33 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 06:00.02 | p3nguin | Yes. Perfectly defined, actually. |
| 06:00.30 | p3nguin | peer = Match incoming requests to a configuration entry using the source IP address and port number. |
| 06:01.19 | p3nguin | It has always been this way for as long as I can remember. |
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| 06:04.47 | kaldemar | think about type=peer and host=dynamic. |
| 06:06.08 | kaldemar | and the book, no matter how good it is, is not a design specification for asterisk. |
| 06:07.51 | p3nguin | As best I can tell, the host=dynamic part allows for a registration, and the registration provides the IP address in which peer does the matching for incoming calls. |
| 06:08.47 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.42) |
| 06:09.06 | kaldemar | the registration is still an incoming request like any other. as the book even says, "incoming requests". |
| 06:09.40 | p3nguin | Okay, you got me there. Now, how does a registration have anything to do with the type? |
| 06:10.09 | kaldemar | the registration matches a device for authentication purposes. |
| 06:10.34 | kaldemar | how else would asterisk know the correct secret? |
| 06:13.19 | p3nguin | I guess I don't understand the question... the secret has to be set in the conf. |
| 06:15.18 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 06:15.22 | p3nguin | I agree that a registration authentication is done with the username and the secret, but that doesn't have anything to do with type being user or peer. It's a completely different transaction, even though the criterion was "incoming request." |
| 06:15.52 | kaldemar | let me rephrase. asterisk must know a correct secret to be able to determine whether a digest reply is valid. to know the correct secret, the request must match a device. and in that phase, there is no IP address nor port when host is dynamic. |
| 06:16.25 | p3nguin | Agreed. Now let us move to the part where type becomes relevant. |
| 06:16.58 | kaldemar | in the device matching part abviously. |
| 06:17.29 | kaldemar | but it is actually close to irrelevant, since both user and peer match on username. |
| 06:17.58 | p3nguin | I feel that we haven't made any progress on this. |
| 06:18.17 | p3nguin | I think we're debating two completely different things. |
| 06:18.46 | ChannelZ | It's all voodoo and cat vomit |
| 06:19.16 | p3nguin | I understand the voodoo part, but cat vomit? |
| 06:20.26 | kaldemar | mathing requests is matching requests, even if registration is not the same transaction as an invite (transaction as specified in SIP). |
| 06:20.37 | p3nguin | The last time I saw a cat vomit, I think it puked up a full-size turd made purely of hair. |
| 06:21.15 | ChannelZ | And there you go |
| 06:21.20 | p3nguin | By that association, Asterisk is a hair turd. :) |
| 06:22.29 | ChannelZ | The docs tend to be |
| 06:22.35 | p3nguin | Hair turd. What a horrible term. |
| 06:23.24 | ChannelZ | registers hairturd.com |
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| 06:56.05 | *** part/#asterisk twitchnln (~twitch@c-98-242-79-16.hsd1.ga.comcast.net) |
| 07:18.36 | SewerCow | is there a better way to learn Asterisk than www.askriskdocs.org? I'm not liking it much. It assumes I have a $600 DAHDI card & 2 servers. |
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| 07:31.42 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 07:38.43 | ChannelZ | Not entirely, it shows SIP and other things as well. |
| 07:39.37 | ChannelZ | I'm writing a primer but probably won't have it done for another couple of weeks. |
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| 08:13.23 | din3sh | having a strange problem, call to a specific country gets hangup after around 3mins systematically |
| 08:13.44 | din3sh | setting rtpholdtimeout= to a high value doesnt do anythg |
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| 09:00.35 | molavy | i have us robotic serial modem , how can setup asterisk use it as voice recording on company closed status |
| 09:01.43 | molavy | any idea? |
| 09:04.08 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 09:23.44 | molavy | there is no idea? |
| 09:25.08 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:e00e:a6c4:c7fa:88b1) |
| 09:30.50 | Henchman21 | http://www.tomshardware.com/forum/18628-42-asterisk-modems-phonecards |
| 09:31.14 | *** join/#asterisk Maliuta_ (~nobusines@kiev.lusan.id.au) |
| 09:31.48 | *** join/#asterisk mintos (~mvaliyav@d58-106-36-120.rdl801.qld.optusnet.com.au) |
| 09:31.51 | Henchman21 | http://www.velocityreviews.com/forums/t233590-p2-asterisk-with-modems-instead-of-phonecards.html |
| 09:34.31 | molavy | i can dial with my linmodem via wvdial, but how can i configure modem work via asterisk |
| 09:34.33 | molavy | ? |
| 09:35.33 | molavy | or configure asterisk to connect to answer incoming call from modem |
| 09:44.24 | bn-7bc | I have a rather strange problem 2 sip friends on the same asterisk books bass issues calling each other: peer a uses ulaw an b uses ills when a call b sound works perfect in both directions, when b calls a no song in any direction. when i switch b t ulaw i have sound no better who places the call (asterisk 10.3.0) |
| 09:45.08 | bn-7bc | for peer b i have tried both groubwire on iOS and several osx sip clients with results above |
| 09:45.20 | bn-7bc | any ideas? |
| 09:49.37 | bn-7bc | I can''t understand why it only works from a to b and tot b to a, I have also tired it with gsm on b same results |
| 09:54.45 | *** join/#asterisk CommaCrazy (~CCK@87.250.37.26) |
| 09:54.51 | CommaCrazy | hey all :) |
| 09:55.58 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 09:56.46 | bn-7bc | paste bin with relevant config : http://pastebin.com/3betGCCe |
| 09:58.19 | CommaCrazy | I got a little problem, every so often, on my Trixbox machine with openvox analog card inserted with 2 lines, I get the prerecorded message telling me that all circuits are busy/try later. When I look at the status of the lines at the web interface it tells me that they are in use... the problem is that asterisk does not clear the line after the call is finished |
| 09:58.43 | CommaCrazy | is there a way to fix this? This happens about every 7 days |
| 10:03.44 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 10:50.49 | markit | hi, I see in many places that the SIP suggested range for port forwarding is UDP 5004:5082. Shouldn't be ONLY 5060 UDP and 5061 TCP for encrypted SIP? |
| 10:57.24 | din3sh | 5060 is the default port |
| 10:57.41 | din3sh | you can change it if you wish |
| 11:01.18 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 11:02.11 | war9407 | Been doing a lot of research, have a SPA3102 w/asterisk 1.8.x, tried different dial plans, different extension parameters, is it possible to pass *67 to block outbound caller ID with astierk and a SPA3102 using an FXO port? |
| 11:03.26 | din3sh | markit: 5061 is default for tls, you can change that also |
| 11:03.52 | markit | din3sh: my question was: why such a wide range is suggested? |
| 11:04.02 | markit | why from 5004 to 5082? |
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| 13:29.52 | cusco | hi |
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| 13:42.23 | rjvvliet | Hi guys, am i coorect that the is no CLI command to find all possible channel variables? |
| 13:44.50 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 13:44.53 | rjvvliet | I'am trying to figure out how to know if i'am in a Attended Transfer (features.conf atxfer => *2) |
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| 14:16.08 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 15:49.35 | Arroyo1010 | Is there anyone who is amongst the living? :) I need to change my SIP trunk provider - Skype. They have some outstanding bug with incoming calls that they can't resolve for a month now. I'm looking for a SIP trunk provider with the following specs: 1 number/3 trunks (maybe more in the future), 5000min/month + charge per minute after that quota. (We will soon go to 10k minutes). Does anyone know a provider that is resonable in pric |
| 15:50.02 | Arroyo1010 | USA/Canada calling only |
| 16:05.50 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 16:05.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 16:07.05 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 16:10.19 | *** part/#asterisk Arroyo1010 (~sdghhf@93-87-163-160.dynamic.isp.telekom.rs) |
| 16:16.56 | pabelanger | ~itsp |
| 16:16.56 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
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| 17:04.16 | rjvvliet | hi guys, i'am trying to use queue channel variables, i have set setqueuevar=yes in queue.conf but when i check ${QUEUECALLS} it is empty, do i first need to call QUEUE_VARIABLES() before they are set? |
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| 17:56.34 | Beltechs | Hello how do I get asterisk to pick up the line when I select the line button. i.e I have a door box connected to a fxs/fxo and I need button line 2 on my phone to pick up the copper line. |
| 17:56.53 | Beltechs | Thanks |
| 18:02.16 | [TK]D-Fender | Beltechs: terminology is vague |
| 18:02.29 | [TK]D-Fender | on "pickup", etc. what this "line" is on what kind of phone, etc |
| 18:02.32 | Beltechs | lol whats up |
| 18:02.41 | Beltechs | yealink |
| 18:02.46 | [TK]D-Fender | what FXS/FXO interface is it? What exact signalling? |
| 18:03.10 | Beltechs | its a digium 410p 2 fxo and 2 fxs... |
| 18:03.26 | [TK]D-Fender | Your device is on ONE part. What kind is it? |
| 18:03.29 | [TK]D-Fender | It isn't BOTH |
| 18:06.35 | din3sh | he has a tdm410 card with 2 fxo and 2 fxs modules |
| 18:06.37 | Beltechs | on my card I have connected 1 copper telco line 1 doorking box and 1 analog telephon |
| 18:06.47 | Beltechs | correct |
| 18:07.20 | Beltechs | I want the analog phone to use the copper line. |
| 18:07.20 | din3sh | whats with the door box? |
| 18:07.31 | Beltechs | intercom |
| 18:07.36 | din3sh | oh ok |
| 18:07.37 | Beltechs | doorking.com |
| 18:08.24 | Beltechs | on my yealing sip phone I want to have line 2 pickup the door box when I select line 2 |
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| 18:08.31 | Beltechs | *k |
| 18:09.41 | din3sh | how does it work? sorry i dont know how the telephony controls the door |
| 18:09.43 | din3sh | :/ |
| 18:10.41 | Beltechs | i just need to pick up the line |
| 18:10.45 | Beltechs | incoming is ok |
| 18:11.55 | din3sh | fxo is ok |
| 18:12.00 | din3sh | fxs not? |
| 18:16.02 | Beltechs | i can dial out, i can receive calls but I need to have the line connect when I pick up the hand set |
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| 19:30.51 | ChannelZ | Beltechs: I think you're going to have to have some extension to dial for your intercom, even if it's one digit. Unless the phone has some sort of autodial function that you can make it trigger as soon as you press the line button or whatever. |
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| 19:35.28 | p3nguin | If you had a Cisco phone and use SCCP on it, chan_sccp has a hotline. |
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| 19:38.10 | ChannelZ | I'm wondering if there is a way to trick the dialplan in the phone but am not at work with mine to try.. like if you had an empty dialplan, does it just not work at all or would it put through a call to no extension |
| 19:38.47 | ChannelZ | I imagine it just does nothing. |
| 19:39.11 | ChannelZ | shrugs and goes back to tearing apart his office to paint |
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| 20:20.38 | [TK]D-Fender | Beltechs: Still around? |
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| 20:35.42 | Beltechs | ya |
| 20:37.54 | [TK]D-Fender | Ok, I had to dip out for a bit. |
| 20:38.05 | [TK]D-Fender | So, clarify exactly what interface your door device is using... |
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| 22:51.22 | a1fa | hey guys. i noticed i cant dial my phones any more.. |
| 22:51.23 | a1fa | # asterisk -vvv Asterisk 1.8.11.0-1digium1~squeeze |
| 22:51.36 | a1fa | if i do sip show peers |
| 22:51.42 | a1fa | my sip peers are available |
| 22:52.26 | a1fa | sip debug does not even show me dialing |
| 22:52.52 | a1fa | 202/202 192.168.1.3 D N 5060 OK (11 ms) |
| 22:54.04 | a1fa | is TK-Defender around |
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| 22:55.23 | jackmcbarn | im looking at tdm400p's on ebay. how can i tell if a card is digium, openvox, chinaroby, or whatever other manufacturer if the listing doesn't say? i know the cards all look different, but i'm not sure what to look for or what a real digium looks like |
| 23:00.35 | a1fa | how can i debug dial plan |
| 23:03.21 | a1fa | huh |
| 23:03.25 | a1fa | its getting caught on timeout |
| 23:03.29 | a1fa | exten => t,n |
| 23:03.32 | a1fa | thats goofy |
| 23:03.37 | a1fa | did anything change in 1.8? |
| 23:04.39 | a1fa | lol its like asking if anyone took a breath today |
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| 23:10.41 | a1fa | lol |
| 23:10.44 | a1fa | i figured it out |
| 23:10.49 | a1fa | the 2nd box was taking calls |
| 23:10.52 | a1fa | friggin strange |
| 23:18.37 | a1fa | somehow, my 2nd pbx booted up and registred to the sip provider as well |
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| 23:26.31 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 23:26.31 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.3.0 (2012/03/29), 1.8.11.0 (2012/03/29), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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