| 00:00.50 | nickgaw | the netbook goes into a low power mode but I can still access it over the network should that be a real problem? |
| 00:01.48 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 00:02.02 | WIMPy | nickgaw: It does what you configure it to do. |
| 00:02.33 | WIMPy | What do you want to do on it anyway? |
| 00:02.54 | nickgaw | I can't find any method in the bios for changing the low power mode so should the second it takes to come up to full power be a problem for handling calls? |
| 00:03.28 | WIMPy | you configure that in your OS. |
| 00:03.46 | WIMPy | And what does low power mode actually mean? |
| 00:14.17 | Mon|A|rch | do you guys know how to compile mp3 support into sox? |
| 00:30.16 | leifmadsen | what distro? |
| 00:32.30 | Mon|A|rch | centos |
| 00:32.50 | leifmadsen | I just googled for, "compile mp3 support into sox centos" |
| 00:32.56 | leifmadsen | gave me info on first 3 links |
| 00:33.03 | Mon|A|rch | i saw those too |
| 00:33.09 | Mon|A|rch | after i asked the question |
| 00:33.13 | Mon|A|rch | like a loser ;_; |
| 00:35.25 | Mon|A|rch | also, are you actually leifmadsen? |
| 00:35.48 | Mon|A|rch | because if you are, your book is fantastic |
| 00:36.15 | leifmadsen | I just wear his skin |
| 00:36.31 | Mon|A|rch | that works too |
| 00:36.47 | leifmadsen | but thanks :) |
| 00:38.03 | *** join/#asterisk timeshell (~timeshell@69-165-165-76.dsl.teksavvy.com) |
| 00:56.14 | phix | haha |
| 00:56.18 | phix | what book? |
| 00:56.29 | ChannelZ | ~book |
| 00:56.29 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 00:57.11 | Mon|A|rch | the latest html version is pretty sweet |
| 00:59.03 | phonebuff | Good Evening. |
| 00:59.52 | phonebuff | Leif: I understand my issues much better from the other day, after putting this aside for awhile I am seeing some of the trees in the forest. |
| 01:00.57 | phonebuff | The patch I had that I thought was for 1.8.18 acctually is for a mix of a couple of different versions, but it was does for chan_sip.c at 1.8.14 and there are a lot of changes between the two. |
| 01:01.34 | phonebuff | So I tried a new approach rather than trying to cross engineer someone else's work. |
| 01:01.59 | phonebuff | IN looking at this URL: https://reviewboard.asterisk.org/r/1976/diff/#index_header |
| 01:03.31 | phonebuff | I would think that support for the Cisco USECALLMANAGER proxy stement made it into 1.10.x and 1.11.x . So I tared up a lot and built a 1.11.2 install but the feature still does not seem to work. |
| 01:04.14 | phonebuff | Am I miss understanmding the notes or did something in the 1.10 trunk not neccessarily make it to the 1.11 base ? |
| 01:05.37 | phonebuff | I really do appreciate your help to date -- |
| 01:08.30 | *** join/#asterisk saint_ (~saint@68.38.56.184) |
| 01:08.32 | saint_ | hi all |
| 01:09.59 | phix | hai saint_! |
| 01:10.48 | saint_ | so.. i'm trying to understand why my asterisk voicemail cannot call back. i call in with a non blocked number, Asterisk says "I'm afraid I don't know who sent this message" ... |
| 01:11.05 | saint_ | Is this feature only good for internal calls, or does it work with external calls too ? |
| 01:11.35 | [TK]D-Fender | saint_: Your call didn't look like what you think it looks like |
| 01:11.43 | [TK]D-Fender | saint_: Go look at the TXT file for it |
| 01:13.13 | saint_ | [TK]D-Fender: which i did.. let me add that this is a google voice number that i am connected to. maybe it does not like the way the caller id shows up ? it shows up as +1xxxxxx@voice.google.com ... Let me try with my other account (voip.ms) and compare the txt file from the voicemail. |
| 01:14.16 | [TK]D-Fender | saint_: tons of junk in the "number" = fail |
| 01:27.29 | saint_ | ok, so when I try with my voip.ms , it says YOU CANNOT REPLY TO THE SENDER BECAUSE HE DOES NOT HAVE A MAILBOX |
| 01:27.43 | saint_ | so this reply function only works for internal calls ? |
| 01:30.22 | rue_house | hello, I have a problem with an mgcp gateway (two of them wont work at once) anyone know if mgcp.conf if limited to one gateway? |
| 01:31.25 | rue_house | is |
| 01:31.52 | *** join/#asterisk j4m3s (~j4m3s@pdpc/supporter/active/j4m3s) |
| 01:32.41 | [TK]D-Fender | saint_: Because you don't understand the scope of it. This is to send a message to that BOX number on YOUR systems |
| 01:32.53 | [TK]D-Fender | saint_: And is assuming that it was left by an internal person. |
| 01:33.13 | [TK]D-Fender | saint_: There is no "transmit a message directly to someone's VM on the PSTN" concept |
| 01:33.57 | saint_ | ha ok, because my current system does it, so I assumed that asterisk would do it too.. never mind then.. thanks |
| 01:34.38 | [TK]D-Fender | Your current PBX can't magically directly drop a voiemail into my CELLPHONE VM box. |
| 01:34.40 | saint_ | i mean.. i m not talking about transmit a mesg to someone's vm on pstn , but calling back the person that left you a voicemail, whether it's an internal number or external number |
| 01:35.03 | saint_ | nah nah.. not to voicemail.. i was justi thinking about calling back the caller. |
| 01:35.04 | [TK]D-Fender | saint_: Show all the backup for your attempt |
| 01:35.17 | [TK]D-Fender | ALL of it |
| 01:35.25 | [TK]D-Fender | the VM txt, full call debug, etc. |
| 01:35.43 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:36.07 | saint_ | it s out of scope. obviously asterisk tries to leave a message in someone's mailbox. i understand it will work only for internal numbers. i was looking for a solution to CALL BACK the person who left a voicemail, whether it's internal or extenal |
| 01:36.37 | rue_house | you just want asterisk to redial the caller who left the message? |
| 01:37.32 | rue_house | there should be caller id data in there, a bit of scripting should be able to get asterisk to call the number back for you |
| 01:37.50 | saint_ | rue_house: yeah. i think i m missing some options in voicemail.conf .. i just found the doc... reading it now.. |
| 01:37.58 | rue_house | ok |
| 01:38.12 | rue_house | do you know anything about mgcp.conf? I'm having troubles |
| 01:38.19 | [TK]D-Fender | saint_: Show the backup |
| 01:38.38 | saint_ | [TK]D-Fender: i don t understand what you are asking by show the backup. |
| 01:39.18 | saint_ | rue_house: i don t deal -yet- with mgcp |
| 01:39.27 | rue_house | drat |
| 01:39.41 | rue_house | cant get more than one gateway working in the config file |
| 01:39.59 | rue_house | I dont think mgcp.conf can handle more than one gateway |
| 01:40.12 | rue_house | that or I'm not understanding the indexing properly |
| 01:40.27 | rue_house | there are no online examples with more than one gateway configured |
| 01:47.07 | [TK]D-Fender | saint_: Show the call. Show the VM TXT. |
| 01:49.55 | rue_house | saint_, I think tk is asking for the text file that pairs with a voicemail that was left |
| 01:50.16 | rue_house | saint_, familiar with where the vm is stored on astersisk? |
| 01:52.49 | j4m3s | if i have an inbound sip call enter a context that dials back out a DAHDI channel, is it possible to set the callerid of the DAHDI call to that of the originating sip call? |
| 01:53.36 | j4m3s | something like Set(DAHDI_CALLERID=${SIPCALLERID}) |
| 01:53.42 | rue_house | j4m3s, I dont think your provider lets you set callerid for an outgoing call |
| 01:54.02 | [TK]D-Fender | j4m3s: It already does that |
| 01:54.20 | rue_house | when you place an outgoing call on an analog line, your provider sets the callerid data that the call recipiant gets |
| 01:54.46 | j4m3s | rue_house, so it's not possible to spoof the callerid of a PSTN call? |
| 01:54.53 | [TK]D-Fender | j4m3s: And not all DAHDI techs allow that concept. Anolog has no "set my callerid" concept |
| 01:54.57 | rue_house | not that I'm aware of |
| 01:55.06 | rue_house | j4m3s, not for outbound calls |
| 01:55.09 | [TK]D-Fender | j4m3s: Not "PSNT", the question is what SIGNALLING |
| 01:55.18 | [TK]D-Fender | j4m3s: PRI, yes. Analog, no. |
| 01:55.20 | j4m3s | fxo |
| 01:55.23 | [TK]D-Fender | no |
| 01:55.33 | [TK]D-Fender | There is no such thing as "setting your caller ID " on analog |
| 01:56.01 | j4m3s | i should prolly know that, but it sure has been a long time since i worked with a FXO line |
| 01:56.45 | j4m3s | [TK]D-Fender, rue_house thanks |
| 01:57.05 | rue_house | [TK]D-Fender, ok NOW will you look at my case? |
| 01:57.41 | [TK]D-Fender | rue_house: So, several weeks later do you actually have something real to show us? |
| 01:57.52 | rue_house | sure, how about mgcp.conf? |
| 01:58.06 | [TK]D-Fender | ... |
| 01:58.09 | [TK]D-Fender | Guess not |
| 01:58.27 | [TK]D-Fender | I'll check back in on it in another few months |
| 01:58.29 | rue_house | http://rueshouse.dyndns.org:82/~asterisk/etc/asterisk/mgcp.conf |
| 01:58.34 | rue_house | thats the config file |
| 01:58.46 | *** join/#asterisk retentiveboy (~retentive@72.54.144.26) |
| 01:58.56 | rue_house | http://rueshouse.dyndns.org:82/~asterisk/var/log/asterisk/messages |
| 01:58.59 | rue_house | there is the problem |
| 01:59.11 | rue_house | Maximum retries exceeded for transaction 6 on [VOG-B] |
| 01:59.51 | *** join/#asterisk AnGrYfUrBy (~AnGrYfUrB@pdpc/supporter/active/angryfurby) |
| 02:00.23 | rue_house | interestingly enough, make a call to a port on VOG-B, it will ring, but shut down as soon as its answered |
| 02:00.24 | *** part/#asterisk AnGrYfUrBy (~AnGrYfUrB@pdpc/supporter/active/angryfurby) |
| 02:00.46 | rue_house | its almost like its a rtp problem |
| 02:06.08 | rue_house | both gateways use the same rtp base port number |
| 02:06.23 | rue_house | which shouldn't matter cause mgcp is supposed to negotiate the port |
| 02:07.13 | rue_house | its one of the few explinations that works for why only one or the other gateway can work at a time |
| 02:07.31 | rue_house | another would be that mgcp.conf just cant handle more than one |
| 02:07.44 | rue_house | the second gateways config is overwriting the first |
| 02:08.26 | rue_house | I went thru the code, but I dont know enough about how the code executes or the data structures to get anything out of it |
| 02:13.15 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 02:13.15 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 02:15.00 | *** part/#asterisk Biblioclasta (~Bibliocla@190.179.194.233) |
| 02:15.12 | rue_house | I think I'm the only person using mgcp tho |
| 02:15.16 | rue_house | for a gateway anyhow |
| 02:15.46 | rue_house | it looks, from what I'v read, like mgcp was written for connecting to serivce providers |
| 02:16.10 | rue_house | which would explain the limitation of 1 |
| 02:18.47 | [TK]D-Fender | And *'s MGCP wasn't written to use with anything except phones. |
| 02:21.52 | rue_house | hu? |
| 02:21.56 | rue_house | hmm |
| 02:22.10 | rue_house | what would be the use of having just one mgcp phne? |
| 02:22.18 | rue_house | station set yo umean yes? |
| 02:22.33 | [TK]D-Fender | no gateways, just phones |
| 02:22.50 | rue_house | so the fast it works with one gateway is amazing |
| 02:22.55 | rue_house | it works fine too |
| 02:23.10 | rue_house | dosn't work with two |
| 02:23.22 | rue_house | do you know of the limitation that might cause that? |
| 02:23.51 | rue_house | is there a proven working example of setting up two sets in mgcp.conf? |
| 02:31.04 | saint_ | back |
| 02:31.35 | rue_house | you might as well just stay logged in |
| 02:32.02 | saint_ | [TK]D-Fender: yes, i am familiar with where the VM is stored. But I found my issue. reply and outgoing calls are option 3/2 and 3/4 in voicemail, and you need apparently to modify the voicemail.conf to add those optiosn, which i did not. so i ll try that first and if it does not work, i'll come back here. |
| 02:32.31 | rue_house | you might as well just stay logged in |
| 02:32.37 | rue_house | :) |
| 02:35.17 | rue_house | anyone here have an application note for the MC34114? My motorola databook is too old (1983) |
| 02:37.38 | saint_ | what does this mean exactly ? WARNING[16014]: chan_sip.c:29269 reload_config: No valid transports available, falling back to 'udp'. |
| 02:38.11 | [TK]D-Fender | It means you didn't explicitly specif which transports to use |
| 02:38.16 | [TK]D-Fender | specify* |
| 02:38.24 | rue_house | udp carries the audio |
| 02:38.48 | saint_ | is this in the sip.conf ? |
| 02:38.49 | rue_house | sip is the connection negotiation |
| 02:39.08 | rue_house | sip needs to know what audio transport mechanism to negotiate with |
| 02:39.35 | saint_ | what's the option in the sip.conf ? |
| 02:40.48 | *** join/#asterisk din3sh (~din3sh@41.136.81.10) |
| 02:41.02 | din3sh | gd mrning all |
| 02:45.11 | rue_house | saint_, how about we switch you to mgcp devices and you can help me :) |
| 02:45.43 | saint_ | rue_house: i m not ready for mgcp yet |
| 02:46.26 | rue_house | its funny really, the M in mgcp is for multi, and asterisk is damned if its gonna let me get more than one going |
| 02:47.08 | jpsharp | mgcp? Masochistic G-d Communications Protocol. |
| 02:47.18 | rue_house | :) |
| 02:47.25 | rue_house | I can get one gateway working but not two |
| 02:47.55 | rue_house | I think that mgcp.conf overwrites bits of its config, so only the last configured gateway works, but nobody can help me |
| 02:48.01 | rue_house | (as usual) |
| 02:48.28 | saint_ | once i'm done with this project, I'm so letting vonage go from my house and installing asterisk with google voice. free incoming / outgoing calls to the USA. Bye bye the $100/month bill ! |
| 02:49.29 | rue_house | isn't a subscriber loop service just $50/mo? |
| 02:49.57 | saint_ | saint_: no clue. I have vonage, and I'm about $100/month |
| 02:50.16 | rue_house | hmm I dont know how to go about making a low-dropout charger |
| 02:50.22 | rue_house | er, wrong channel |
| 02:50.33 | jpsharp | $100 for vonage? Does that come with a personal assistant? |
| 02:50.52 | saint_ | jpsharp: nah, it comes with international, and some virtual numbers in foreign countries |
| 02:51.02 | saint_ | that's what bumps up my bill |
| 02:51.38 | jpsharp | Ah. Yes, that'll do it. |
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| 02:56.08 | saint_ | is there any way to have the dial by name read the recorded name of the voicemail, instead of spelling the name configured in the voicemail.conf file ? |
| 02:56.56 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 02:57.25 | [TK]D-Fender | saint_: it does |
| 03:02.24 | ChannelZ | Unless they didn't record one right |
| 03:03.59 | [TK]D-Fender | Water is wet |
| 03:06.05 | saint_ | [TK]D-Fender: oh yeah, got it to work, sweet. Asterisk is awesome. |
| 03:07.13 | saint_ | gotta to. good night all. thanks for the help. |
| 03:07.19 | saint_ | s/to/go |
| 03:12.56 | din3sh | there are 2 new fields in CDR table * 1.8, namely peeraccount & linkedid. I have added these fields in the CDR table but not data is being recorded |
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| 03:29.15 | phix | [TK]D-Fender: a cow goes m00 |
| 03:30.21 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 03:32.59 | saint_ | [TK]D-Fender: every hole has an edge. |
| 03:33.09 | saint_ | good night all. |
| 03:33.41 | DaPrivateer | OK, so if anyone can assist ... I'm trying to set up TLS with a Polycom 670 and keep getting this error: Problem setting up ssl connection: error:1409441B:SSL routines:SSL3_READ_BYTES:tlsv1 alert decrypt error |
| 03:33.51 | rue_house | can anyone help me with a mgcp problem? my second gateway wont work |
| 03:34.52 | [TK]D-Fender | rue_house: Virtually nobody uses MGCP with *. Go post on the mailing list. |
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| 03:37.52 | joako | When do the asterisk RPMĀ“s normally rebuild? |
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| 03:39.20 | rue_house | nobody uses newbridge T1 channelbanks either |
| 03:39.25 | rue_house | EXCEPT ME |
| 03:39.57 | rue_house | wish I could know what the state sequence it was getting on a hangup was so I could code it in |
| 03:45.40 | jpsharp | Ohman, I haven't touched newbridge equipment since 1998. |
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| 05:39.15 | rue_house | it runs my house great |
| 05:41.23 | rue_house | can anyone help me with a mgcp problem? my second gateway wont work |
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| 06:25.14 | phix | hey I am having issues with a TDM4100p, I need to press the buttons longer for them to be picked up, anyway to adjust that? |
| 06:25.36 | phix | any button I press on the keypad I want it to registry regardless on how long I push it down for |
| 06:26.15 | kaldemar | that's not how it works really. |
| 06:26.42 | ChannelZ | Are you talking about FXS channels, from analog phones? |
| 06:28.47 | kaldemar | phix: try relaxdtmf=yes in chan_dahdi.conf if it helps. |
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| 06:33.30 | mattsl | How precise is the voicemail maxsecs setting? |
| 06:34.02 | phix | kaldemar: ok |
| 06:34.42 | mattsl | I've been having an issue where voicemail would drop after 10-60 secs. I think I fixed it, but it dropped the last one at 550 secs with a maxsecs=600 |
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| 07:37.45 | morfin | can't i set CDR variables from gosub? |
| 07:38.29 | WIMPy | The gosub from the Dial()? |
| 07:38.42 | morfin | yes |
| 07:38.58 | WIMPy | IIRC you're on the wrong channel there. |
| 07:39.03 | morfin | maybe return value |
| 07:39.17 | morfin | it's like some function i think |
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| 07:50.41 | kaldemar | morfin: all CDR fields are not settable in the first place. you can set variables on the calling channel in U() with func MASTER_CHANNEL. |
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| 08:07.47 | Rico29 | does anybody here use XML with aastra sip phones (57i, 673xi) ? |
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| 08:14.20 | PbxMan | morning |
| 08:14.39 | WIMPy | Cool. Here as well. ;-) |
| 08:19.09 | Rico29 | hi :) |
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| 08:55.49 | mattsl | If anyone would be so kind as to lend a hand, I'm still trying to track down a problem I had earlier with voicemail dropping the call. I'm going to stay online here, but have to get at least a little sleep. I've posted details and logs here: http://forums.asterisk.org/viewtopic.php?f=1&t=85487 Thanks in advance if anyone has any clue as to what might be happening! |
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| 09:02.56 | ludoooo | hi |
| 09:02.58 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.52.185) |
| 09:03.25 | ludoooo | i'm a problem on asterisk CLI> because extension not found in context 'default'. |
| 09:07.59 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 09:09.29 | kaldemar | ludoooo: still? show your sip.conf and a sip debug with verbosity of an incoming call. |
| 09:10.04 | *** join/#asterisk apb1963__ (~apb1963@174.134.117.244) |
| 09:11.51 | ChannelZ | That's because the extension you dialed wasn't found in the context called 'default'. |
| 09:11.52 | ChannelZ | NEXT! |
| 09:12.55 | kaldemar | there's probably a peer matching problem among with other config problems. |
| 09:14.21 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 09:14.24 | ludoooo | kaldemar wait please |
| 09:14.28 | bombev | hi all |
| 09:14.52 | ChannelZ | listens to MOH |
| 09:15.43 | *** join/#asterisk timahvo1 (~rogue@197.182.75.140) |
| 09:16.11 | ludoooo | here my configuration : http://pastebin.com/PJumypte |
| 09:17.46 | ChannelZ | did you mean to say 192.168.1.203 for the hostname of t38modem-options ? |
| 09:18.45 | ChannelZ | because your call is coming from .203 but your only defined peer is set to .202 so it's matching nothing and going into the default context |
| 09:21.18 | ludoooo | On the 202, the configuration is the same |
| 09:22.00 | ChannelZ | huh? |
| 09:22.16 | ludoooo | I have 2 VM (202 and 203) |
| 09:22.36 | ChannelZ | ok so you're showing us half the config |
| 09:22.51 | ChannelZ | neither part of which match your problem |
| 09:24.23 | ChannelZ | The fact is, your have a call coming from 192.168.100.203:6061 which seems to match no peer based on your sip.conf, so it's going to the dialplan context "default"... and there is no matching extension in that context (which its self probably doesn't exist.) |
| 09:24.23 | ludoooo | I have 2 VM which allows my tests. The two have t38modem with asterisk installed. I try to send a fax from 203 to 202 through t38modem. |
| 09:24.54 | ludoooo | Mhmm ok |
| 09:26.13 | ChannelZ | you also seem to Dial() SIP/t28modem0 which doesn't exist either so I'm not really sure what you're intending to accomplish |
| 09:26.26 | ludoooo | But how can I point the fax to [t38modem0]? |
| 09:26.57 | ludoooo | i'm sorry for my english |
| 09:27.39 | ludoooo | i'm french |
| 09:28.14 | ChannelZ | set the host= IP to be, presumably, .203 which is where it's actually coming from.' |
| 09:28.35 | ChannelZ | and the port seems to be 6061 not 6060 |
| 09:28.47 | ludoooo | ok |
| 09:29.00 | *** join/#asterisk schmidts (~schmidts@77.116.37.252.wireless.dyn.drei.com) |
| 09:29.06 | schmidts | good morning |
| 09:29.31 | ChannelZ | The config you showed defines a peer called [t38modem0] with the IP 192.168.100.202:6060. But the console you pasted seems to show the call coming from 192.168.100.203:6061 |
| 09:29.49 | *** join/#asterisk ghghz (~ton@kluonis.kvb.lt) |
| 09:32.13 | ludoooo | ok |
| 09:32.23 | ChannelZ | Assuming you fix the IP/port to match that of the failed incoming call, it then is going to try to Dial(SIP/t28modem/...) but you have no peer called t28modem0 so that will then fail.. and the syntax doesn't make sense either |
| 09:33.30 | ChannelZ | It's Dial(SIP/extension@provider) or Dial(SIP/peername/extension) |
| 09:33.47 | ghghz | Hey. Does anyone know any provider which allows to change callerid ? |
| 09:33.52 | ludoooo | yes, I just change |
| 09:33.56 | ludoooo | thx |
| 09:34.47 | ChannelZ | ghghz: Vitelity does. |
| 09:35.07 | schmidts | ghghz first it depends on the country you want this, and second there are two different way of doing this. setting a user provided number should be possible at every provider, setting a network provider on your own mostly needs some legal stuff before |
| 09:35.10 | ludoooo | How can he know that it will point to [t38modem0] not [general] ? |
| 09:35.35 | ghghz | schmidts: second case |
| 09:35.41 | ChannelZ | ludoooo: because it has to match the peer, either by name or by IP/port (which depends on the peer type) |
| 09:35.55 | ChannelZ | for incoming calls that is |
| 09:36.37 | ChannelZ | If it can't find a match, it winds up applying being an "anonymous call" and will go into the context specified under [general] |
| 09:37.07 | schmidts | ghghz then it depends even more in which country you are |
| 09:37.30 | ghghz | understand |
| 09:39.06 | ludoooo | ok |
| 09:41.17 | ghghz | ChannelZ: are you using vitelity? |
| 09:42.00 | ChannelZ | yes (US) |
| 09:46.32 | bombev | schmidts gmogning |
| 09:46.35 | bombev | morning :) |
| 09:47.09 | ChannelZ | someone needs some over-the-counter wake-up drugs |
| 09:54.37 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.52.185) |
| 09:55.21 | fling | ChannelZ: what is the best way to record all the calls automagically? |
| 09:56.02 | schmidts | morning bombev did you found any solution for your problem yesterday? |
| 09:58.02 | schmidts | fling using monitor from extensions.conf? |
| 09:58.26 | fling | schmidts: thanks! :] |
| 09:59.07 | fling | schmidts: may I pass left call leg into left wav channel and right call leg into right wav channel? |
| 10:00.04 | *** join/#asterisk deo (~deo@58.71.19.178) |
| 10:00.14 | *** join/#asterisk ghost75 (~trechber@dslb-092-075-061-251.pools.arcor-ip.net) |
| 10:00.58 | schmidts | fling it depends on how you configure the monitor app, imho its default to generate two wav files |
| 10:05.11 | ghost75 | its recording only one channel |
| 10:09.30 | fling | ghost75: so I can't record two channels? |
| 10:12.33 | kaldemar | fling: Monitor makes two files, MixMonitor makes one mixed file. |
| 10:14.53 | *** join/#asterisk barbosa2 (~juliano.b@186.226.71.6) |
| 10:18.55 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 10:19.26 | dym | Hey people. I seem to experience a massive qualityloss when transforming a pdf into "SendFax" capable tif files. Is there any good advice on how to work around this? |
| 10:20.23 | dym | I use: "gs -q -dNOPAUSE -dBATCH -dSAFER -sDEVICE=tiffg3 -sOutputFile=output.tif input.pdf" for the transformation |
| 10:21.09 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 10:33.46 | schmidts | dym it depends on the input image but let my check my settings |
| 10:37.32 | schmidts | dym thats how i do this: s -q -dNOPAUSE -dBATCH -dSAFER -sDEVICE=tiffg32d -sPAPERSIZE=a4 -sOutputFile=".$this->faxdir."fax.tif /usr/local/bin/autorotate.ps -f ".$this->faxdir."*_converted.ps; |
| 10:38.48 | schmidts | this autorotate.ps script only rotates the image if necessary |
| 10:42.36 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 10:45.36 | *** join/#asterisk jacekowski (jacekowski@jacekowski.org) |
| 10:48.11 | din3sh | anybody's implemented email2fax using hylafax? |
| 10:48.51 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:49.15 | schmidts | din3sh i have both, email2fax using hylafax and also using sendfax from asterisk direct |
| 10:52.38 | din3sh | how do you configure email2fax? |
| 10:53.48 | din3sh | pain in the ass to get this working |
| 10:55.57 | schmidts | din3sh i am using exim4 with smtp authentication and in the exim conf you can configure a script when a mail is received from the smtp server, this script gets executed. in this script you have to get the mime attachment of the mail, convert it as mentioned above and then hand it over to hylafax (sendfax) or directly to asterisk using a callfile |
| 11:01.15 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:06.39 | jacekowski | i've got elastix/freepbx with asterisk 1.8 |
| 11:07.02 | jacekowski | and every time a call fails with any cause code it gives caller "all circuits are busy now" message |
| 11:07.27 | jacekowski | i can change the dialplan and everything, but i'm jut wondering, are there any better messages around |
| 11:09.18 | din3sh | schmidts:how do you use sendfax with asterisk directly |
| 11:10.49 | Bladerunner05 | Hello any italian who use fastweb ? |
| 11:15.33 | jacekowski | is there a recording list somewhere? |
| 11:19.49 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.46.255) |
| 11:20.46 | schmidts | din3sh just convert to a tif image and then use a callfile |
| 11:22.34 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 11:26.04 | din3sh | oki |
| 11:26.13 | din3sh | I have a problem with my E1 |
| 11:26.24 | din3sh | whereby local pstn numbers can call in |
| 11:26.38 | din3sh | but international numbers sometimes can get through |
| 11:26.48 | din3sh | other times cannot get through |
| 11:26.56 | din3sh | what might be the problem? |
| 11:27.12 | WIMPy | What makes you think it has something to do with you? |
| 11:27.39 | schmidts | din3sh sounds more like a problem of your carrier, not having enough capacity for international calls |
| 11:28.57 | din3sh | the client had a nortel on which the so called DID worked all the time, i have jst shifted to an Asterisk based system |
| 11:29.03 | din3sh | the call does not come in |
| 11:29.30 | din3sh | when 3mins earlier the call can come in |
| 11:30.04 | WIMPy | What exactely does "not come in" mean? If it means what it says it can't be you. |
| 11:30.10 | din3sh | I have suggested that the problem might be with the provider, the client doesnt want to hear that |
| 11:30.36 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:30.42 | din3sh | say you call a DID at any random time, the phone rings |
| 11:30.56 | schmidts | din3sh ignoring facts didnt solve any problems ;) what kind of busy signaling does the caller hear? is it a normal busy or a fast busy? |
| 11:31.11 | din3sh | you try to call the same DID say 5mins later, you have a message from provider saying the line is not available. |
| 11:31.25 | din3sh | strange since other DID calls to the system continue to work |
| 11:31.33 | WIMPy | Look at pri debug. |
| 11:31.33 | din3sh | and there is no dahdi alarm on the line |
| 11:32.10 | din3sh | schmidts: what is the difference between normal and fast busy? |
| 11:32.38 | WIMPy | fast busy = congestion |
| 11:32.50 | WIMPy | Or some other error. |
| 11:33.07 | schmidts | WIMPy thanks :P |
| 11:33.21 | din3sh | fast would mean that the call does reach the * box? |
| 11:33.59 | WIMPy | To find that out, look at pri debug. |
| 11:34.08 | *** join/#asterisk jsjc (~Adium@186.153.78.188.dynamic.jazztel.es) |
| 11:34.43 | din3sh | what would a pri log for an incoming call to * look like |
| 11:34.49 | schmidts | din3sh either it reaches * and get some error code return or it doesnt even reach it, as WIMPy says you have to take a look at the pri debug, if you might see an incoming call and some kind of an error going back |
| 11:34.50 | din3sh | sorry for noob question |
| 11:35.21 | din3sh | ok |
| 11:35.31 | WIMPy | You will see if it comes in :-) message type SETUP. |
| 11:35.49 | din3sh | in this case i got an annoucement not from * but from the provider sayin "the number you have called is not available" |
| 11:36.40 | WIMPy | That can happen if the call came for a non-existant extension. |
| 11:36.57 | din3sh | oh ok |
| 11:37.07 | WIMPy | But you should see that in the normal console. |
| 11:37.18 | din3sh | but the thing is some callers get the DID, other callers cannot |
| 11:37.32 | din3sh | means that there is a matching for the DID |
| 11:37.40 | WIMPy | And it's definitely the same extension? |
| 11:39.32 | din3sh | yes |
| 11:39.33 | WIMPy | BTW: "can happen" wasn;t quite correct. That's what *will* happen for calls to unconfigured extensions. |
| 11:39.56 | schmidts | WIMPy this also could happen if somewhere on the way to his * is wrong, cause of using another ss7 path or a wrong configured CIC |
| 11:39.56 | din3sh | to make matters worst |
| 11:40.24 | din3sh | i have frequent HDLC errors on span 1 |
| 11:40.38 | WIMPy | If it use.... |
| 11:40.43 | WIMPy | That's an explanation. |
| 11:41.09 | WIMPy | You have interrupt issues, wrong configuration or a bad line or bad hardware. |
| 11:41.55 | WIMPy | Now that I think about it. Didn't I write that twice yesterday? |
| 11:42.06 | din3sh | PRI got event: HDLC Abort (6) on D-channel of span 1 |
| 11:42.22 | din3sh | chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 |
| 11:42.24 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 11:42.33 | din3sh | yes WIMPy |
| 11:42.44 | din3sh | now the client going nuts |
| 11:42.57 | din3sh | the issue reaching boiling point |
| 11:43.04 | WIMPy | If that happens more than a few times a week, you've got a problem. |
| 11:43.19 | din3sh | the HDLC error |
| 11:43.25 | din3sh | its daily |
| 11:43.29 | din3sh | contineously |
| 11:43.31 | WIMPy | yes |
| 11:43.45 | WIMPy | Then something is broken. |
| 11:44.00 | din3sh | actually i have this setup : PSTN----->ASterisk-------->Nortel |
| 11:44.31 | WIMPy | You just put Asterisk in between? |
| 11:44.36 | din3sh | I have HDLC errors on both spans |
| 11:44.38 | din3sh | yes |
| 11:44.50 | WIMPy | First check your timing configuration. |
| 11:45.18 | din3sh | thats why the client is unhappy, prior to installing asterisk in between, they didnt have any issue with the nortel |
| 11:45.52 | WIMPy | Yes, we already know it's b0rked. Now go and find out why. |
| 11:51.04 | din3sh | my chan_dahdi.conf --> http://pastebin.com/kvQW5Rqx |
| 11:51.16 | din3sh | can you plz take a look if its ok |
| 11:51.33 | WIMPy | dahdi/system.conf does the timing. |
| 11:52.25 | din3sh | going nuts :S |
| 11:52.33 | *** join/#asterisk Cubber (~ronny@cpe-74-71-250-35.twcny.res.rr.com) |
| 11:52.45 | WIMPy | But that dialplan configuration will surely give you some trouble as well. |
| 11:53.43 | WIMPy | Always set the dialplan parameters to "unknown" unless you really know what you're doing. |
| 11:54.47 | din3sh | pridialplan=national? |
| 11:54.53 | din3sh | set both to unknown? |
| 11:55.09 | din3sh | system.conf ----> http://pastebin.com/ZwmAZ513 |
| 11:55.19 | WIMPy | yes |
| 11:56.08 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 11:56.24 | WIMPy | Timing looks ok. |
| 11:56.41 | WIMPy | But I would expect that you have crc on that line. |
| 11:57.16 | WIMPy | That is you might want to add ,crc4 after the hdb3. |
| 11:58.43 | din3sh | I tried, the hdlc didnt stop |
| 11:59.21 | WIMPy | What does dahdi_test give you? |
| 12:02.20 | din3sh | I ran dahdi test from 9pm to 4am |
| 12:02.22 | din3sh | Best: 100.000% -- Worst: 99.862% -- Average: 99.995729% |
| 12:02.47 | din3sh | Worst: 99.862% not good? |
| 12:03.17 | WIMPy | Not the optimum, but not bad enough to explain the errors. |
| 12:04.10 | WIMPy | Have you tried to replace cables? |
| 12:04.52 | din3sh | yes to no avail |
| 12:05.30 | din3sh | I replaced the cable from ASterisk to Nokia DNT2Mi CAt5 cable to Cat7 |
| 12:05.33 | din3sh | it got worst |
| 12:05.51 | din3sh | i mean the errors, lots of alarm on the line |
| 12:06.16 | WIMPy | Cat3 is fine :-) |
| 12:10.24 | din3sh | lol |
| 12:10.43 | WIMPy | Maybe the line termination? 75 vs 120 Ohm. You don;t have any baluns there anywhere, do you? |
| 12:12.21 | din3sh | no |
| 12:12.25 | din3sh | no baluns |
| 12:12.48 | din3sh | intersting to note that before shifting to the current * box |
| 12:13.10 | din3sh | i had a test box, a desktop pc |
| 12:13.27 | din3sh | with zaptel and asterisk 1.4 instead of dahdi and asterisk 1.8 now |
| 12:13.39 | din3sh | the zaptel and asterisk 1.4 didnt fire up any hdlc |
| 12:13.54 | din3sh | i didnt have any call connection issues |
| 12:14.43 | WIMPy | That sounds as if there's some interrupt processing going wrong despite dahdi_test giving acceptable results. |
| 12:14.48 | din3sh | the new elements are the server, new longer cables, dahdi asterisk 1.8 |
| 12:15.41 | WIMPy | I'd try another board. |
| 12:15.45 | din3sh | server top shows processing only at around 20-30% |
| 12:16.18 | din3sh | current one is a new HP DL380 G6 |
| 12:16.34 | din3sh | single quad, 8gb, raid1 300gb |
| 12:16.36 | WIMPy | Depending on kernel configuration that doesn't mean much. It might not show interrupt activity. |
| 12:18.29 | din3sh | cat /proc/interrupts shows wcte12xp0 |
| 12:18.30 | din3sh | <PROTECTED> |
| 12:19.02 | din3sh | is it good practice to have irqbalance? |
| 12:19.53 | WIMPy | What card is that? |
| 12:20.24 | WIMPy | In an edge case i found that dahdi actually worked better when it was restricted to only one CPU. |
| 12:20.41 | WIMPy | But under normal conditions it shouldn't matter. |
| 12:21.24 | WIMPy | If you want to really push it, set the dahdi IRQ to the last core and all others to the others. |
| 12:21.55 | WIMPy | But I wouldn't expect any change from doing so. |
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| 12:26.51 | *** join/#asterisk Cubber (~ronny@cpe-74-71-250-35.twcny.res.rr.com) |
| 12:28.03 | din3sh | digium TE220 with echo cancel |
| 12:31.23 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-scmbsssuwfldlaxu) |
| 12:31.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:34.02 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:34.58 | WIMPy | Ah, PCIe. Hard to get shared IRQs there. :-) |
| 12:35.51 | WIMPy | Hmm. |
| 12:36.03 | WIMPy | wonders if it could be a power management issue. |
| 12:37.20 | WIMPy | has had some issues with ASPM. |
| 12:37.44 | WIMPy | With other kind of hardware that is. |
| 12:44.51 | *** join/#asterisk rokjan (~jj2@static-190-181-29-206.acelerate.net) |
| 12:45.26 | fling | how to increase outgoing sound volume for a sip peer? |
| 12:45.43 | fling | I want to be louder :P |
| 12:46.00 | WIMPy | Configure your terminal. |
| 12:46.08 | fling | WIMPy: umm? |
| 12:46.17 | kaldemar | turn up the volume in your phone. |
| 12:46.30 | kaldemar | or use func VOLUME, which is worse. |
| 12:47.00 | WIMPy | In my experience it's really bad. |
| 12:47.01 | fling | kaldemar: I'm using softphone and I see in envy24control my sound is loud enough |
| 12:47.07 | fling | kaldemar: bus sip provider is bad |
| 12:47.21 | fling | s/bus/but |
| 12:47.49 | WIMPy | Sounds unlikely that they (can) change your volume. |
| 12:52.02 | fling | WIMPy: ok |
| 12:55.32 | fling | I'm creating a simple dialplan with call recording using Monitor |
| 12:55.43 | fling | is this looks good? > http://dpaste.com/889968/ |
| 12:56.05 | leifmadsen | don't use SetVar() |
| 12:56.07 | leifmadsen | just use Set() |
| 12:56.19 | fling | is fixing SetVar |
| 12:56.26 | leifmadsen | first line, by my preference, should be a NoOp() |
| 12:56.37 | leifmadsen | otherwise, looks fairly clean |
| 12:57.19 | din3sh | WIMPy:WIMPy has had some issues with ASPM.? |
| 12:57.24 | din3sh | like what? |
| 12:57.41 | fling | leifmadsen: first line in which context? |
| 12:57.46 | WIMPy | Like things not working any more. |
| 12:58.03 | leifmadsen | oh, and _skype, you look fine there, but in the future realize the X, N, and Z are special chars, so you'd need something like _[n]ew[z]eala[n]d for words that contain those letters |
| 12:58.18 | ludoooo | hi |
| 12:58.27 | leifmadsen | fling: and context that has an extension that uses same |
| 12:58.33 | leifmadsen | exten => s,1,NoOp() |
| 12:58.38 | leifmadsen | <PROTECTED> |
| 12:58.38 | leifmadsen | etc |
| 12:58.52 | fling | oh! thanks |
| 12:58.52 | din3sh | mine is working but erractically |
| 12:58.55 | leifmadsen | I don't like the first priority of an extension (unless it is only 1 priority) to have functionality |
| 12:59.06 | leifmadsen | that's a style issue though, nothing is technically wrong with how you did it |
| 12:59.17 | WIMPy | din3sh: Yes, that's the kind of issue I had. |
| 13:01.20 | fling | leifmadsen: no x/n/z in 'skype#' :p |
| 13:01.38 | leifmadsen | <leifmadsen> oh, and _skype, you look fine there, but in the future realize... |
| 13:02.43 | fling | leifmadsen: sorry, thanks! |
| 13:04.49 | fling | leifmadsen: http://dpaste.com/889973/ |
| 13:06.52 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 13:08.55 | fling | leifmadsen: and my ${EXTEN} become 's' :p |
| 13:09.00 | fling | -- Called SIP/s@zadarma |
| 13:09.09 | fling | how to fix it? |
| 13:11.49 | leifmadsen | well ${EXTEN} is the extension that is being executed |
| 13:12.22 | WIMPy | Where is that call coming from? |
| 13:12.36 | leifmadsen | [outgoing] |
| 13:12.51 | fling | yes, I'm dialing |
| 13:12.57 | leifmadsen | if you are expecting it to be the contents of the first pattern match, then you need to save that to a channel var, then use that channel var in the [outgoing] context |
| 13:13.10 | fling | hmm hmm |
| 13:13.14 | leifmadsen | exten => _+7.,1,NoOp() |
| 13:13.21 | leifmadsen | same => n,Set(CalledNumber=${EXTEN}) |
| 13:13.30 | leifmadsen | same => n,Goto(outgoing,s,1) |
| 13:13.32 | leifmadsen | <PROTECTED> |
| 13:13.41 | leifmadsen | then replace ${EXTEN} in your Dial() with ${CalledNumber} |
| 13:13.54 | fling | simple, thanks, almost done |
| 13:13.58 | WIMPy | Or don't use s. |
| 13:14.14 | leifmadsen | honestly I would make [outgoing] a subroutine and pass in the data as an argument |
| 13:14.21 | leifmadsen | rather than a Goto() |
| 13:14.38 | leifmadsen | GoSub(outgoing,s,1(${EXTEN})) |
| 13:14.51 | WIMPy | Or just replace s with ${EXTEN} in the Goto. |
| 13:15.17 | leifmadsen | then you need the same pattern match in multiple places |
| 13:15.42 | fling | leifmadsen: please help me editing it :p http://dpaste.com/889973/duplicate/ |
| 13:16.25 | WIMPy | Or just one catchall. |
| 13:16.38 | leifmadsen | http://dpaste.com/889997/ |
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| 13:19.23 | fling | leifmadsen: thanks, it works |
| 13:23.31 | fling | the problem in this line 'same => n,Set(CALLFILENAME=incoming/${TIMESTAMP:0:4}/${TIMESTAMP:4:2}/${TIMESTAMP}-${CALLERIDNUM})' that all the vars are empty :p where to get the values? |
| 13:41.02 | kaldemar | fling: use functions STRFTIME and CALLERID. |
| 13:41.32 | kaldemar | fling: those variables have been deprecated and removed ages ago. they are not set unless you set them yourself in the dialplan. |
| 13:42.14 | fling | kaldemar: thanks |
| 13:42.27 | *** join/#asterisk din3sh (~din3sh@196.20.243.199) |
| 13:44.00 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 13:45.00 | *** join/#asterisk rbd_ (~rbd@cpe-173-095-180-095.nc.res.rr.com) |
| 13:45.48 | *** join/#asterisk kontinuity (~kontinuit@122.172.225.46) |
| 13:45.54 | kontinuity | hi there people |
| 13:46.18 | kontinuity | I am having issues with queues and cdr |
| 13:46.41 | kontinuity | apparently the updatecdr=yes in queues.conf doesn't work, it generates duplicate CDR |
| 13:46.49 | rbd_ | asterisk 1.6: I have SET statements for vars like "Command4Text" in a callfile that I need to refer to in my extensions.conf .... I have a loop counter var called LoopCount ... would something like "${Command${LoopCount}Text}" work in extensions.conf ? |
| 13:46.59 | kontinuity | I am on Asterisk 1.6.2.12 |
| 13:47.16 | kontinuity | any help would be much appreciated |
| 13:49.39 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 13:54.10 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 13:55.19 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 13:55.20 | *** join/#asterisk serafie (~erin@nat/digium/x-cohljnlcjszhobnf) |
| 13:56.56 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-scmbsssuwfldlaxu) |
| 14:00.21 | *** join/#asterisk w9sh (~chatzilla@64.238.96.125) |
| 14:01.41 | fling | kaldemar: what about this? same => n,Set(CALLFILENAME=outgoing/${STRFTIME(${EPOCH},GMT+7,%Y.%m.%d)}/${STRFTIME(${EPOCH},GMT+7,%Y.%m.%d-%H:%M:%S)}_${CALLERID(num)}_${ARG1}) |
| 14:02.05 | *** join/#asterisk acedia (~rage@unaffiliated/ffs) |
| 14:02.05 | fling | kaldemar: GMT+7 is not working, it works like GMT-7 :p |
| 14:05.20 | kaldemar | fling: the time zone should be something in /usr/share/zoneinfo, e.g. US/Alaska. |
| 14:05.22 | fling | kaldemar: just as with regular date. replaced with a right timezone and now it works |
| 14:05.37 | *** join/#asterisk ke-esc (~ke-esc@155.229.78.130) |
| 14:06.00 | fling | I need to put this date string into a separate variable firstā¦ |
| 14:06.08 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:06.54 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:06.54 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:08.43 | ke-esc | Hi all- I'm planning an asterisk installation for a small business. The only actual office is in CT, but there are employees throughout the US. There is a T1 for phone trunks, and a T3 to the internet. Remote workers connect via SIP. Would you imagine any latency/quality issues for employees on the west coast? If so, what do people do to rectify that? |
| 14:11.12 | leifmadsen | ke-esc: the latency issues across the continental US isn't that much |
| 14:11.13 | pabelanger | ke-esc: better control of the network |
| 14:11.21 | leifmadsen | nothing that is going to be too much of an issue |
| 14:11.32 | leifmadsen | this assumes a relatively sane path across the internet |
| 14:11.59 | pabelanger | Right, but manage how much bandwidth each worker requires |
| 14:12.01 | leifmadsen | alternatively, you create an MPLS tunnel across to your server, but if there isn't a single location for multple people, that isn't really feasible |
| 14:12.50 | ke-esc | we actually have an mpls currently, but not all the employees are on it (only the engineers, not sales people since they travel more than not) |
| 14:14.13 | fling | kaldemar: now when I call I first say 'The call may be recorded' :D |
| 14:16.13 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 14:16.57 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 14:17.22 | *** join/#asterisk Quest (~syncsys@119.152.247.140) |
| 14:17.44 | Quest | is there a way to send faxes without a fax machine / modem nor with a paid service like efax.com? |
| 14:19.26 | [TK]D-Fender | Quest, You direct from * or with things connecting through it |
| 14:19.30 | [TK]D-Fender | you can* |
| 14:19.53 | leifmadsen | yes. |
| 14:20.11 | Quest | sorry? |
| 14:20.14 | leifmadsen | you can use someone like flowroute and send faxes via T.38 |
| 14:20.36 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 14:20.50 | Quest | consider there is not fax machine or modem. so a fax can be sent by t.38? |
| 14:20.59 | Katty | has put dehydrated worms on the feeder this morning. |
| 14:21.30 | Quest | [TK]D-Fender, leifmadsen ? |
| 14:22.32 | *** join/#asterisk deo (~deo@112.198.90.72) |
| 14:22.37 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-kpzsiivjsivfwvca) |
| 14:22.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:22.52 | schmidts | Katty and i guess this brokes your asterisk? |
| 14:23.20 | [TK]D-Fender | Quest, Yes |
| 14:23.44 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:24.09 | Katty | schmidts: naturally. |
| 14:24.14 | Quest | [TK]D-Fender, wow.. how is that possible? |
| 14:25.01 | [TK]D-Fender | Quest, I don't understand your confusion..... |
| 14:25.11 | Quest | leifmadsen, [TK]D-Fender how it can be send? if i have t.38 licence (which is free and can be obtained easily) |
| 14:25.25 | schmidts | Katty as usual ;) if i didnt know it better i would say you are one of my kids, whenever i told them to not broke anything, they will brake it for sure :P |
| 14:25.29 | [TK]D-Fender | Quest, With *'s internal fax apps, etc |
| 14:25.46 | Quest | [TK]D-Fender, leifmadsen look. consider there is no fax machine or modem. now how a fax can be sent.? |
| 14:25.50 | [TK]D-Fender | ... |
| 14:26.00 | Quest | [TK]D-Fender, by a computer that has a software in it? |
| 14:26.12 | Quest | the computer being connected to a telephone line? |
| 14:26.13 | [TK]D-Fender | Quest, You're in #asterisk ... don't you HAVE Asterisk? |
| 14:26.16 | Quest | right? |
| 14:26.20 | Quest | [TK]D-Fender, yes. |
| 14:26.32 | [TK]D-Fender | Quest, What are YOU talking about? You sound like you're asking if a fax can happen out of NOTHING |
| 14:26.55 | Quest | i have asterisk. i have a DiD link to another country. i want to send faxes to that country. |
| 14:26.56 | [TK]D-Fender | Quest, ASTERISK can take to the PSTN. And thus it's passing audio, etc |
| 14:27.10 | Quest | [TK]D-Fender, well what are the requirements of sending a fax? |
| 14:27.19 | leifmadsen | if there is no fax, use a scanner and send the pdf |
| 14:27.38 | Quest | leifmadsen, where to send the pdf ? |
| 14:27.41 | leifmadsen | Quest: ofps.oreilly.com has the latest Asterisk book up. Read the fax chapter for ideas on how to accomplish what you need. |
| 14:27.43 | [TK]D-Fender | Quest, It IS the software. So yes if your * can reach the PSTN you can send faxes using T.38 passthrough from other software, or directly with it's dialplan fax apps |
| 14:27.48 | leifmadsen | Quest: already documented, so check that out |
| 14:29.39 | Quest | precisely, asterisk can send faxes to any number in the world through a t.38 (without needing a fax machine / modem) like by scanning the document and sending it as .pdf through asterisk. correct? |
| 14:30.23 | *** join/#asterisk thecoda (~kevin@027b5f4b.bb.sky.com) |
| 14:30.39 | [TK]D-Fender | Quest, * can send faxes through things it connects to |
| 14:31.04 | Quest | [TK]D-Fender, "through things it connect to" ? for example? |
| 14:31.20 | thecoda | Got an unusual problem with caller id, looking for advice |
| 14:31.52 | [TK]D-Fender | Quest, You seem to be incredibly vague about how you reach the phone network |
| 14:32.20 | [TK]D-Fender | Quest, Asterisk has to be able to reach the REAL WORLD and you are leaving this factor our of your question every time. |
| 14:32.57 | Quest | [TK]D-Fender, well there are phones connected to asterisk of-course. but do you mean that a document be converted to .pdf . given to asterisk and then to a phone and then the it sends it as fax? |
| 14:33.16 | [TK]D-Fender | Quest, Connected to Asterisk how? |
| 14:33.23 | Quest | Did. |
| 14:33.25 | Quest | by did |
| 14:33.28 | [TK]D-Fender | NO |
| 14:33.28 | thecoda | I have an inbound PSTN line, ${CALLERID}, ${CALLERIDNUM}, ${CALLERIDNAME} all come up empty when I try to log them to the console, but the extension I pass the inbound call to is correctly showing caller ID. What gives? |
| 14:33.31 | [TK]D-Fender | A DID is a number |
| 14:33.36 | Katty | [TK]D-Fender: are you being nice this morning? |
| 14:33.42 | [TK]D-Fender | it is not a WIRE type, or signalling |
| 14:33.46 | Katty | [TK]D-Fender: not giving asterisk a bad name, right? |
| 14:33.52 | Quest | [TK]D-Fender, ok. |
| 14:33.56 | *** join/#asterisk moos3 (~moos3@pool-72-95-122-23.ptldme.east.myfairpoint.net) |
| 14:33.59 | Quest | [TK]D-Fender, so what do i need? |
| 14:34.03 | ke-esc | I think its time for the PFM response :) |
| 14:34.09 | [TK]D-Fender | Quest, You don't seem to understand what you have. |
| 14:34.15 | Katty | Quest: try not to take [TK]D-Fender too seriously. |
| 14:34.20 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 14:34.21 | Katty | Quest: he's a bit rough around the edges. |
| 14:34.33 | moos3 | is there a way to have a counter and increment it ? basically i want to make the menu repeat 5 times if no input hang up |
| 14:34.36 | Quest | nevermind.. I just need to know what i need? |
| 14:34.58 | Katty | Quest: not unlike a very cranky child that needs a nap. |
| 14:35.03 | [TK]D-Fender | Quest, .... ther are dozens of technologies to get to the PSTN. |
| 14:35.24 | Quest | like? |
| 14:35.32 | [TK]D-Fender | Quest, It's like asking me "get me transportation". Do i know if a BIKE will do? Or a car? Do you need an aircraft carrier? |
| 14:35.32 | Katty | analog lines. |
| 14:35.33 | Katty | T1 |
| 14:35.42 | [TK]D-Fender | ITSP |
| 14:35.45 | Katty | "sip trunk" the sales folk like to call it |
| 14:35.45 | [TK]D-Fender | Cellular. |
| 14:35.59 | [TK]D-Fender | ISDN-BRI |
| 14:36.02 | [TK]D-Fender | The list goes on. |
| 14:36.05 | [TK]D-Fender | PHEON LINES> |
| 14:36.14 | Katty | you know i read a wikipedia article about buzz words last night. |
| 14:36.20 | Quest | yes. I can affoard a phone linE:) |
| 14:36.22 | Katty | it was pretty amusing. |
| 14:36.45 | [TK]D-Fender | Quest, Asterisk is software. It doesn't have a jack you can pulg a wire into. So what ARE you looking to do? |
| 14:36.55 | thecoda | So nobody has a clue about my caller ID mystery? |
| 14:37.25 | [TK]D-Fender | Quest, I can send faxes directly over the internet via an ITSP. There. I have no "line" of my own. |
| 14:37.33 | Quest | well. basically what are the requrements of sending a fax. the fax machine converts the document to a fax readable and sendable format. so if asterisk can do the same. theres no need to have a fax machine... right? |
| 14:37.45 | [TK]D-Fender | <thecoda> I have an inbound PSTN line, ${CALLERID}, ${CALLERIDNUM}, ${CALLERIDNAME} all come up empty when I try to log them to the console, but the extension I pass the inbound call to is correctly showing caller ID. What gives?<- these are Asterisk 1.0 variables and have been dead for over half a decade. |
| 14:37.58 | [TK]D-Fender | thecoda, "core show function CALLERID" |
| 14:38.25 | [TK]D-Fender | Quest, it depends on how you plan on reaching the PSTN. |
| 14:38.27 | Katty | [TK]D-Fender: you sir need a hug and an espresso. |
| 14:38.30 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 14:38.34 | [TK]D-Fender | Quest, Answer that and we'll be able to tell you |
| 14:38.46 | schmidts | Hey asterisk guys, i want to hook up an SDM-64 to an asterisk server do you think this will work? |
| 14:38.55 | Katty | schmidts: hey. and girls, mister. |
| 14:39.11 | schmidts | Katty i think a cup of hot milk would be a better idea ;) |
| 14:39.12 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.56.193) |
| 14:39.14 | Quest | [TK]D-Fender, we have a did from Alkaraiz. we do calls in U.S by that. we have an asterisk server running too. what else do we need> |
| 14:39.28 | Katty | schmidts: at this point? i think scotch is the only thing that will work |
| 14:39.40 | [TK]D-Fender | Quest, A DID is just a phone number. That doesn't say how it is DELIVERED |
| 14:39.47 | schmidts | katty +1 |
| 14:39.53 | WIMPy | Katty: Is guy male for you? I had the impression it's used in a neutral way. |
| 14:40.06 | Katty | WIMPy: i never took it as gender neutral. |
| 14:40.11 | Katty | WIMPy: but it's entirely possible some do |
| 14:40.16 | Quest | [TK]D-Fender, how the pstn can be reach at U.S . because we want the "sender" of fax to be in U.S .(despite the real sender will be out side u.s. here.) |
| 14:40.18 | schmidts | btw katty i am sure you know the female word for nerd? |
| 14:40.31 | [TK]D-Fender | Quest, Answer the question. |
| 14:40.45 | Quest | which one/ |
| 14:40.52 | Quest | how to reach pstn? |
| 14:40.55 | Katty | schmidts: not particularly. |
| 14:40.56 | [TK]D-Fender | Quest, How is this DID delivered |
| 14:41.03 | Katty | schmidts: but i'm sure you will tell me here soon! |
| 14:41.09 | schmidts | quest whisper the answer is 47 |
| 14:41.14 | Quest | [TK]D-Fender, .. dont know.. |
| 14:41.20 | Quest | iam not that technical |
| 14:41.21 | [TK]D-Fender | facepalms |
| 14:41.25 | schmidts | Katty i also dont know it thats why i asked :D |
| 14:41.38 | Quest | Alkaraiz is the company |
| 14:41.47 | Katty | Quest: that's part of the problem. |
| 14:41.53 | Katty | Quest: you need your technical person in here. |
| 14:42.10 | Katty | schmidts: we should consult the google! |
| 14:42.15 | Quest | [TK]D-Fender, Katty by pbx i should say, if thats what you are asking |
| 14:42.26 | [TK]D-Fender | Quest, I can't tell if that is a REAL PHONE LINE with a hardware card in your server or if they are an ITSP and Asterisk is speaking to it using VoIP, etc |
| 14:42.29 | schmidts | katty i dont think aunt google knows the answer |
| 14:42.41 | [TK]D-Fender | Quest, But you DO have an Asterisk server that already talks to the real phone network? |
| 14:42.50 | Quest | [TK]D-Fender, its voip . yes |
| 14:43.10 | Katty | schmidts: google says nerdette |
| 14:43.15 | thecoda | Beautiful, and callerID is fully armed and operational |
| 14:43.16 | [TK]D-Fender | Quest, VOIP was the answer. I'm sure at some point you could tell what PROTOCOL it is actually using.... that is going to be very important |
| 14:43.31 | Quest | ok.. |
| 14:43.39 | Quest | [TK]D-Fender, so what now? |
| 14:43.52 | Katty | schmidts: but nerd has such a negative connotation :< |
| 14:43.55 | leifmadsen | thecoda: use the CALLERID() function, not those outdated channel variables |
| 14:43.56 | [TK]D-Fender | Quest, So Asterisk has dialplan apps for this already and is documented |
| 14:44.03 | leifmadsen | ${CALLERID(all)} for example |
| 14:44.13 | Katty | good morning leif. |
| 14:44.13 | schmidts | Katty really? what kind of? |
| 14:44.22 | Quest | [TK]D-Fender, theres not fax machine / modem needed? |
| 14:44.24 | leifmadsen | Katty: good morning! I have a br0ked asterisk box I need you to fix |
| 14:44.36 | Katty | leifmadsen: brt! |
| 14:44.36 | Quest | [TK]D-Fender, just the dialpalm is needed? |
| 14:44.40 | *** join/#asterisk scubes13 (~scubes13@39.sub-70-193-15.myvzw.com) |
| 14:44.41 | [TK]D-Fender | Quest, Not if you are using a VOIP PROVIDER |
| 14:44.47 | [TK]D-Fender | Quest, THEY are the modem |
| 14:44.52 | Katty | oh. dialpalm. i like that. |
| 14:44.57 | Quest | [TK]D-Fender, hmm. |
| 14:45.01 | thecoda | Yeah, I didn't spot that callerid was overloaded as both a var and a func |
| 14:45.03 | [TK]D-Fender | Quest, Asterisk talks to the PSTN through them. Faxes are audio on the PSTN |
| 14:45.05 | Katty | infobot: dialpalm |
| 14:45.13 | leifmadsen | :D |
| 14:45.15 | Quest | [TK]D-Fender, oooh |
| 14:45.21 | thecoda | Easily spotted when you know what to look for |
| 14:45.26 | Katty | infobot: dialpalm is facepalming, but with phone stuffs. |
| 14:45.26 | infobot | okay, Katty |
| 14:45.34 | leifmadsen | Katty: nice! |
| 14:45.53 | bombev | hi guys |
| 14:46.03 | Katty | herro. |
| 14:46.14 | bombev | what does that mean: WARNING[22037] codec_g729a.c: out of G.729 decoder licenses |
| 14:46.15 | bombev | <PROTECTED> |
| 14:46.32 | [TK]D-Fender | bombev, "Out of licenses". How much clearer can that be? |
| 14:46.41 | Katty | [TK]D-Fender: manners. sir. |
| 14:46.50 | [TK]D-Fender | bombev, You had licenses. You ran out. You need more. You don't have more. Call = dead. |
| 14:46.53 | schmidts | katty more scotch!! |
| 14:46.53 | Quest | [TK]D-Fender, is that correct (ignoring voip and asterisk or did for a moment) that fax document needs to be converted to a fax readable format by a fax machine or modem and only a fax machine / modem can send a fax? |
| 14:46.59 | Katty | schmidts: indeed. |
| 14:47.02 | Katty | [TK]D-Fender: more scotch. immediately. |
| 14:47.12 | Katty | [TK]D-Fender: put the keyboard down and back away slowly mister cranky pants. |
| 14:47.14 | [TK]D-Fender | Katty, You're not helping. |
| 14:47.42 | Katty | [TK]D-Fender: and neither is your cranky bad attitude. |
| 14:47.49 | Katty | [TK]D-Fender: you are driving people away. |
| 14:47.50 | schmidts | well girls and guys have a nice morning/day/evening *waves* |
| 14:47.51 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 14:47.54 | Katty | [TK]D-Fender: so why don' you think about that. |
| 14:48.08 | [TK]D-Fender | Katty, No, they are here, and learning and getting answers from someone. |
| 14:48.11 | Quest | [TK]D-Fender, what is the function of fax machine? |
| 14:48.31 | [TK]D-Fender | Quest, To send and receive faxes\ |
| 14:49.04 | Quest | [TK]D-Fender, what do they do that cannot be done be a computer if connected to a phone line with a dialup modem? |
| 14:49.17 | Quest | done by * |
| 14:49.54 | Quest | [TK]D-Fender, can a fax be send by a computer connected to a phone line? |
| 14:49.59 | [TK]D-Fender | Quest, Where are you going to shove physical paper into your computer? You didn't say you had a scanner, so if you have a PAPER you want sent as a fax I guess you aren't going to get anywhere, are you? |
| 14:50.03 | bombev | [TK]D-Fender well I have g729 license: 0/0 encoders/decoders of 2 licensed channels are currently in use |
| 14:50.03 | bombev | Licenses Found: |
| 14:50.03 | bombev | File: G729-JL3MWAC****.lic |
| 14:50.09 | Quest | [TK]D-Fender, .pdf ...... |
| 14:50.14 | Greenlight | Anyone know what with calls made over DAHDI (E1 ISDN30's) sometimes asteriks says "Call XXX is making progress passing it to..." and sometimes it says "Call XXX is ringing...." |
| 14:50.23 | [TK]D-Fender | Quest, you want to receive a fax? But I wan't it PAPER in my hand. You didn't say you had a printer either. |
| 14:50.27 | Quest | [TK]D-Fender, the computer scans paper and converts to pdf and sends by phone line |
| 14:50.34 | [TK]D-Fender | bombev, "not enough" <- |
| 14:50.38 | moos3 | anyone have informatino how to do a counter in a dailplan |
| 14:50.45 | Quest | [TK]D-Fender, only talking about sending at the moment |
| 14:50.48 | WIMPy | Greenlight: Depends on where you're calling. |
| 14:50.53 | bombev | [TK]D-Fender well what should i do? how come not enough? |
| 14:51.09 | Quest | [TK]D-Fender, so is possible? |
| 14:51.24 | [TK]D-Fender | Quest, a scanner is a separate peripheral for a computer... not everone has one. |
| 14:51.29 | Greenlight | WIMPy: So kinda equivelant to "early media" or just a ringing signal ? |
| 14:51.51 | *** join/#asterisk ludoo43333 (~IceChat9@lns-bzn-51f-81-56-143-71.adsl.proxad.net) |
| 14:51.52 | [TK]D-Fender | bombev, You are using more channels than you have licenses for |
| 14:51.53 | Quest | [TK]D-Fender, ignore scanner. imply that the document to be sent is alrady in .pdf form. |
| 14:52.17 | morfin | hmm |
| 14:52.19 | [TK]D-Fender | Quest, you should already have enough to answer this yourself. |
| 14:52.30 | Quest | [TK]D-Fender, just this last question |
| 14:52.41 | Quest | [TK]D-Fender, what do you say? |
| 14:52.45 | [TK]D-Fender | moos3, What are you counting? From what point? Only within a call? |
| 14:53.07 | [TK]D-Fender | Quest, Asterisk can send faxes. I've answered this 10 times in the past half an hour |
| 14:53.14 | moos3 | the first time I repeat the menu |
| 14:53.17 | [TK]D-Fender | Quest, DIRECT from * out whatever you have connected to it |
| 14:53.19 | Quest | [TK]D-Fender, i said. ignore asterisk |
| 14:53.20 | WIMPy | Greenlight: Normally it's always erly media. there are many call states, all until release are optional. |
| 14:53.23 | bombev | [TK]D-Fender axa, thats why perhaps, because between extenions g729 works great |
| 14:53.40 | Quest | (ignoring voip and asterisk or did for a moment) |
| 14:53.46 | bombev | but when I try to call via sip trunk this is what happens |
| 14:53.50 | Quest | [TK]D-Fender, ^ |
| 14:54.06 | moos3 | [TK]D-Fender basically they enter the menu it plays every thing waits 3 seconds then does a GoToIF Count < 3 replay else hangup |
| 14:54.07 | Greenlight | Yea, with SIP it's always been early media and I *always* get "XXX is making progress...". But with DAHDI seems to be quite random, and I don't understand why |
| 14:54.20 | [TK]D-Fender | Quest, Your question makes no sense. Rephrase it |
| 14:54.23 | Quest | [TK]D-Fender, assume i dont have asterisk. i only have a computer, phone modem and a phone line. can i send fax with that by a pdf? |
| 14:54.32 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 14:54.37 | [TK]D-Fender | moos3, Set a variable. Increment it. Done. |
| 14:54.44 | *** join/#asterisk mandla (~mmapeo@41.223.142.96) |
| 14:54.46 | Quest | [TK]D-Fender, no asterisk. no did. nothing... |
| 14:55.03 | Gugge | Quest: without some software no, with some other software than asterisk .... i guess someone else in the whole world made software that can send a fax :) |
| 14:55.06 | [TK]D-Fender | Quest, Ever her of WinFax? there are OTHER softwares for sending fax and have for decades. |
| 14:55.15 | Quest | sure.. with some software installed on pc |
| 14:55.31 | Quest | hm.. |
| 14:55.33 | moos3 | [TK]D-Fender how do you increment it is really my question |
| 14:55.36 | [TK]D-Fender | Quest, Would you believe that Asterisk is the only software capable of sending faxes from a computer? |
| 14:55.42 | [TK]D-Fender | moos3, MATH. |
| 14:55.51 | [TK]D-Fender | moos3, Dialplan expressions 101 |
| 14:55.59 | moos3 | so $COUNTER=$COUNTER++ ? |
| 14:56.05 | Quest | i meant any software that can be installed on windows... i mean are there such software that can send faxes? |
| 14:56.12 | [TK]D-Fender | moos3, "dialplan expressions" <- Book |
| 14:56.13 | [TK]D-Fender | ~book |
| 14:56.14 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:56.22 | Greenlight | Windows has a built in fax printer driver iirc |
| 14:56.25 | Gugge | moos3: the function INC maybe :) |
| 14:56.34 | WIMPy | Greenlight: It might be the called PBX for example. |
| 14:56.42 | [TK]D-Fender | Quest, there are hundreds and thousands of fax software on this planet already.... |
| 14:56.48 | thecoda | I'm waiting for the 4th edition, currently running on asterisk 11... |
| 14:56.48 | leifmadsen | Set(COUNTER=${INC(COUNTER)}) |
| 14:56.54 | moos3 | leifmadsen nice thanks |
| 14:56.56 | leifmadsen | thecoda: ofps.oreilly.com |
| 14:57.00 | leifmadsen | you can review it |
| 14:57.02 | thecoda | nods |
| 14:57.08 | Quest | [TK]D-Fender, Gugge so fax is just a voice call like thing. it has no special formate? if theres a fax machine on the other side. it will print out the fax. no matter send by a software in a pc connected to a phone.. right? |
| 14:57.17 | WIMPy | Greenlight: chan_dahdi for example is quite noisy on incomming calls. |
| 14:57.21 | moos3 | [TK]D-Fender thansk |
| 14:57.40 | Quest | [TK]D-Fender, and in that senario. i do need a t.38 line? |
| 14:57.41 | Gugge | Quest: you send sound through a phone line yes |
| 14:57.44 | [TK]D-Fender | Quest, The format IS the sound on the line. |
| 14:57.49 | Quest | t.38 is a line or certifcate? |
| 14:58.00 | [TK]D-Fender | NEITHER. |
| 14:58.04 | [TK]D-Fender | ~T.38 |
| 14:58.04 | infobot | t.38 is, like, see t38 |
| 14:58.08 | [TK]D-Fender | ~t38 |
| 14:58.08 | infobot | [t38] see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 14:58.26 | Greenlight | WIMPy: Ok thanks; I'm trying to get a reliable way to tell when a call *actually* starts ringing, via the AMI. The normal channel state of "ringing" is a lie, so i've had to trigger a custom message on the ".. making progress" which works great for SIP, but seems DAHDI is going to be more troublesome |
| 14:58.33 | [TK]D-Fender | Quest, It is a protocol used over VoIP to allow reliable faxing over the internet |
| 14:58.42 | [TK]D-Fender | Quest, adnd your PROVIDER has to support it |
| 14:59.37 | Quest | [TK]D-Fender, oh so (now implying that i do have asterisk , a did from Alkaraiz, and voip) it implies alkaraiz would be having t.38 |
| 14:59.46 | WIMPy | Greenlight: The ringing message should do exactely that. |
| 14:59.57 | [TK]D-Fender | Quest, No. It implies that they had BETTER support it. |
| 15:00.03 | Quest | ok |
| 15:00.11 | WIMPy | Greenlight: What kind of issue do you have there? |
| 15:00.14 | Greenlight | WIMPy: No, as soon as you place a call Asterisk classes it as "ringing" |
| 15:00.17 | [TK]D-Fender | Quest, Because sending as straight audio is going to have a very high failure rate |
| 15:00.18 | Quest | so i can just start sending faxes just as i do calls? |
| 15:00.28 | Greenlight | When in real life, the other phone doesn't start to ring for 2-8 seconds |
| 15:00.35 | Greenlight | And that is a LONG time |
| 15:00.51 | WIMPy | Greenlight: Sounds like a bug. |
| 15:00.52 | [TK]D-Fender | Quest, Yes, for the eleventh time. |
| 15:00.53 | Greenlight | So, the Asterisk ringing message, like the cake, is a lie |
| 15:01.44 | Greenlight | When I place a call there is no intermediate channel state withing Asterisk, just Down -> Ringing -> Up ? |
| 15:02.02 | Greenlight | I presumed this was a well known issue |
| 15:02.03 | [TK]D-Fender | Greenlight, Up = answered. |
| 15:02.09 | Greenlight | No shit |
| 15:02.12 | Quest | [TK]D-Fender, so nice of you |
| 15:02.17 | [TK]D-Fender | Greenlight, Down is "nothing", so all that leaves is "ringing" |
| 15:02.37 | Greenlight | So what do we call that state inbetween a call being placed and it actually ringing |
| 15:02.57 | Greenlight | Asterisk puts the channel in a state of "ringing" (as per AMI events) |
| 15:03.03 | Quest | [TK]D-Fender, i will brb |
| 15:03.03 | [TK]D-Fender | Greenlight, "milliseconds" |
| 15:03.14 | Greenlight | It's not milliseconds |
| 15:03.30 | [TK]D-Fender | Greenlight, They're plural..... exactly how many ... YMMV |
| 15:03.38 | Greenlight | You call a mobile phone from Asterisk over a SIP carrier, and it's ringing in milliseconds |
| 15:03.48 | WIMPy | Can't easily check with dahdi, but with LCR I get the ringing state exactely when it starts ringing. |
| 15:03.59 | Greenlight | LCR ? |
| 15:04.44 | WIMPy | Linux Call Router. I use that to connect Asterisk to the ISDN instead of DAHDI. |
| 15:04.54 | Greenlight | I see |
| 15:05.07 | Greenlight | So, before it starts ringing, what state is the channel in ? |
| 15:05.11 | Greenlight | Like |
| 15:05.17 | WIMPy | And there is the channelstat 3 "Dialing" for the time in between. |
| 15:05.18 | Greenlight | I place a call to a mobile |
| 15:06.07 | Greenlight | Hmm - That channelstate would make sense, but it never happens looking at the AMI events, over any of a number of SIP or ISDN carriers i've tried |
| 15:06.42 | Greenlight | I thought this was a well known "issue"; there's a few posts online and even a ticket to try and get more events added |
| 15:06.44 | WIMPy | I don't get that, either. |
| 15:07.22 | WIMPy | I'm pretty sure I used it with dahdi as well, way back then. |
| 15:07.27 | Greenlight | Most SIP carriers i've used are a second or two slower than PSTN phones to start ringing |
| 15:07.44 | Greenlight | And of course with mobile phones it's even worse |
| 15:08.11 | Greenlight | ANd we've got Ofcom industry regulations to adhere to in terms of how long calls are to be ringing for at minimum so need to be exact |
| 15:08.58 | Greenlight | At present Asterisk tells me state of "Down" (0) then "Ring" (4) in the same microsecond |
| 15:09.09 | Greenlight | The next update I get with the default codebase is "Up" |
| 15:09.22 | WIMPy | ring != ringing |
| 15:10.08 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:10.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:10.20 | Greenlight | Hmm |
| 15:10.26 | Greenlight | What's the difference between the two states ? |
| 15:10.32 | WIMPy | as always the documentation isn't really helpful, but I thing ring means incoming ring or incoming overlap receive. |
| 15:10.41 | WIMPy | think |
| 15:11.02 | Greenlight | Okay |
| 15:11.17 | Greenlight | goes away to look over some logs and check what messages he gets through |
| 15:12.22 | Greenlight | Wow |
| 15:12.39 | WIMPy | hm. Don't even have AMI enabled on the box using dahdi. |
| 15:12.45 | Greenlight | Okay, this has certainly changed from 1.6 when I first implenmented this, unless I was just being braindead at the time |
| 15:12.56 | Greenlight | Looking at a log of a call from earlier today |
| 15:13.32 | WIMPy | I would have guessed I did it with 1.4. |
| 15:13.58 | WIMPy | So you get a correct ringing state? |
| 15:14.01 | Greenlight | Down (0ms) -> Ring (0ms) -> Ringing (1523ms) -> Progres (1523ms) -> Up (10233ms) |
| 15:14.20 | Greenlight | That "Progress" event is my own custom code in Asterisk to fire that event |
| 15:14.21 | *** join/#asterisk shadar (~eugene@37.113.202.81) |
| 15:14.43 | Greenlight | But it's triggerd the same time as a "Ringing" event |
| 15:15.10 | WIMPy | "but"? |
| 15:15.24 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 15:15.38 | Greenlight | Perhaps I've just assumed Ring and Ringing were the same when I initially codeed this |
| 15:15.57 | Greenlight | Cause looking at it now, that Ringing message is all I need to accurately tell when it actually starts to ring |
| 15:16.32 | *** join/#asterisk scubes13 (~scubes13@69.26.198.2) |
| 15:16.54 | Greenlight | Would save me having to patch Asterisk with my own code each release |
| 15:17.07 | Greenlight | Wonder if I get that on DAHDI, would guess so... |
| 15:22.22 | WIMPy | I even get state dialing. |
| 15:22.35 | Greenlight | Via your LCR ? |
| 15:22.48 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-205-242.red.bezeqint.net) |
| 15:23.03 | WIMPy | No, dahdi. |
| 15:23.20 | Greenlight | Oh - Am just getting logged onto my box with DAHDI to have a check |
| 15:23.40 | Greenlight | So you go: Down -> Ring -> Dialing -> Ringing - Up ? |
| 15:25.43 | Greenlight | 220371272013-01-24 15:24:24.2431GLAI::ChannelEvent_NewStateChannel has changed state. Channel: DAHDI/i1/XXXXXXXXX-100d, State: 3, StateDesc: Dialing. |
| 15:25.48 | WIMPy | No. Newchannel:Rsrvd -> DAHDIChannel -> Newtate:Dialing -> Dial -> Newstate:Ringing |
| 15:25.52 | Greenlight | Yup - I get Dialling on dahdi too |
| 15:26.24 | Greenlight | Ah, perhaps DADHI uses Dialling instead of Ring |
| 15:27.09 | thecoda | hmm, can't find much on using OSLEC that dates from after 2009 |
| 15:27.10 | WIMPy | Ring doesn't seem to make sense there anyway. |
| 15:27.27 | thecoda | Given that it's now in the kernel, I suspect many of the guides are now wrong |
| 15:27.36 | Greenlight | I get Rsvd as well actually |
| 15:28.03 | Greenlight | Yea, I only had SIP to develop on so not really had a good look at DAHDI until it wasn't behaving correctly as I noticed earlier |
| 15:28.03 | [TK]D-Fender | thecoda, The process hasn't changed in a while. |
| 15:28.29 | thecoda | Any pointers to a 2.9 source tree can't be right :P |
| 15:28.48 | WIMPy | So it's just a case of don't guess, read. And it magically just works. |
| 15:29.17 | Greenlight | I'm fairly sure I never used to get that "Ringing" message, which is why I implemented my own Progress event. I hope it's an update to Asterisk that sorted that, rather than me using a different SIP carrier |
| 15:29.36 | Greenlight | I know SIP can send both Ringing messages (180 ? ) and Early Media (183 ?) |
| 15:30.06 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-205-242.red.bezeqint.net) |
| 15:30.23 | Greenlight | Maybe my old carrier that I used when developing and testing originally didn't send the ringing message |
| 15:30.29 | thecoda | Can anyone advise as to why my phones take a lot longer to start ringing via dahdi than when connected directly |
| 15:30.34 | WIMPy | Don;t know what ITSPs send out, other that I wouldn;t pu too much faith in it. But As stated: I think I used to do ecactely that way back in the Asterisk 1.4 times. |
| 15:30.45 | *** join/#asterisk navaismo (~navaismo@189.191.207.103) |
| 15:31.24 | Greenlight | Then I guess it's more related to how it interprets SIP messages and translates that into channel states |
| 15:31.29 | WIMPy | With zaptel, that is. |
| 15:31.38 | thecoda | I'm going pstn - (dahdi) -> asterisk - (sip) -> gigaset - (dect) -> handset |
| 15:32.00 | thecoda | pstn -> gigaset -> handset is *much* faster |
| 15:32.17 | WIMPy | thecoda: What kind of pstn/dahdi? |
| 15:32.27 | thecoda | UK/analogue |
| 15:32.42 | Greenlight | I think what I'll do then is leave my custom patch to trigger a state of "Progress" in there, but also catch the state of "Ringing", and ignore the "Progress" event if "Ringing" has been triggered. Should work with DAHDI and most SIP carriers I'd have thought |
| 15:32.46 | WIMPy | Probably waiting for caller ID. |
| 15:34.04 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 15:34.23 | Greenlight | Damn that's annoying. It seems I *dont* get "Ringing" on DAHDI at all |
| 15:34.47 | Greenlight | RSvd -> Dialling -> Progress -> Up |
| 15:35.03 | WIMPy | What are you calling. |
| 15:35.08 | Greenlight | A UK landline |
| 15:35.16 | Greenlight | Residential landline on the PSTN |
| 15:35.29 | WIMPy | As said earlier (in another context) everything up to hangup is optional. |
| 15:35.42 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:35.42 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:35.48 | Greenlight | Yea, how bloody annoying ^^ |
| 15:35.53 | WIMPy | That probably doesn;t answer without ringing, I guess. |
| 15:36.04 | thecoda | If I call from my mobile with both asterisk and another phone on my landline then I hear one ring on my mobile, then the fixed phone sounds along with the second ring, then asterisk along with the third |
| 15:37.10 | WIMPy | thecoda: Must be waiting for caller ID. |
| 15:37.12 | thecoda | I know it's normal to hold back on one ring to support SMS, but I'm not sure what's causing the delay of the second one |
| 15:37.36 | WIMPy | the second call. |
| 15:37.38 | thecoda | No, UK uses polarity reversal before the ring for caller ID |
| 15:37.39 | Greenlight | Yea, you could be right - it seems to answer almost immedialty after my "Progress" event, gonna dig through some more calls. Hopefully there's enough there to devise a solution that'll let me accurately tell when a line starts to ring |
| 15:37.58 | WIMPy | One call from pstn to asterisk and then another call from asterisk to your cordless phone. |
| 15:38.46 | thecoda | possibly, but the cordless is SIP, so I'd expect ringing to be handled out of band |
| 15:38.54 | *** join/#asterisk elico (~Thunderbi@109.64.164.192) |
| 15:39.01 | WIMPy | It surely is. |
| 15:39.03 | thecoda | unless it's an oddity of the gigaset base |
| 15:39.37 | WIMPy | Look at the *CLI to see where the delay actually happens. |
| 15:39.53 | WIMPy | (or possibly delayS) |
| 15:40.44 | thecoda | Starting simple switch on 'DAHDI/1-1' the instant I hear dialling on my mobile |
| 15:40.49 | fling | how to run wav->flac encoding after HangUp right in dialplan? |
| 15:40.56 | thecoda | then two rings before the sip phone sounds |
| 15:41.17 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 15:41.23 | thecoda | I'll try with a soft phone, rule out the gigaset/dect/whatever |
| 15:41.25 | jmls | afternoon all |
| 15:41.39 | WIMPy | Simple switch is only for phones AFAIK. So that looks like a configuration issue to me. |
| 15:41.45 | jmls | is it possible to limit the automixmon option to certain users only ? |
| 15:41.56 | jmls | or do I have to run a macro from features.conf that checks ? |
| 15:41.57 | thecoda | <PROTECTED> |
| 15:41.57 | thecoda | <PROTECTED> |
| 15:41.59 | thecoda | <PROTECTED> |
| 15:42.00 | thecoda | <PROTECTED> |
| 15:42.02 | thecoda | <PROTECTED> |
| 15:42.03 | thecoda | <PROTECTED> |
| 15:42.05 | thecoda | <PROTECTED> |
| 15:42.06 | thecoda | <PROTECTED> |
| 15:42.07 | thecoda | <PROTECTED> |
| 15:42.08 | thecoda | <PROTECTED> |
| 15:42.10 | thecoda | <PROTECTED> |
| 15:42.10 | WIMPy | ~pb |
| 15:42.10 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 15:42.15 | thecoda | This is an inbound pstn call |
| 15:42.58 | WIMPy | So it might even look for destinctive rings. |
| 15:43.22 | WIMPy | If you don;t have such a thing, you surely want to disable it. |
| 15:44.00 | thecoda | Now I have to discover how :) |
| 15:45.19 | thecoda | This is only a home system |
| 15:45.43 | thecoda | and when I say *only* I mean that such niggles have to pass the rather extreme "wife test" |
| 15:48.04 | *** join/#asterisk l2trace99 (~Thunderbi@rrcs-71-43-25-66.se.biz.rr.com) |
| 15:50.01 | Greenlight | Ahh the WAF, often the stumbling block for many a pet project |
| 15:50.12 | [TK]D-Fender | thecoda, Your delay is DAHDI waiting for callerid. |
| 15:50.37 | [TK]D-Fender | thecoda, a directly connected phone will start ringing immediately and show the CID as it arrives, but not slow down the ringing. Asterisk doesn't work that way |
| 15:50.43 | *** join/#asterisk jrad (~jrad@173.11.144.149) |
| 15:50.49 | [TK]D-Fender | thecoda, So either disabled callerID support, or live with the delay |
| 15:51.58 | thecoda | I know that CID is sent after the first ring in the US. But in the UK it's signalled by polarity reversal before any ring |
| 15:53.57 | *** part/#asterisk ghost75 (~trechber@dslb-092-075-061-251.pools.arcor-ip.net) |
| 15:54.06 | [TK]D-Fender | thecoda, Also seems to be checking ringing patterns "multi-ring" |
| 15:54.15 | [TK]D-Fender | thecoda, If you care about that... then that's an additional delay |
| 15:54.22 | thecoda | Yeah, that's what I'm looking to disable |
| 15:54.39 | [TK]D-Fender | thecoda, that's "dring" normally in chan_dahdi.conf |
| 15:54.47 | din3sh | c/lear |
| 15:54.48 | thecoda | usedistinctiveringdetection=no in chan_dahdi.conf didn't seem to help |
| 15:59.46 | *** join/#asterisk deo (~deo@112.198.90.6) |
| 16:07.42 | *** join/#asterisk ghost75 (~trechber@dslb-092-075-061-251.pools.arcor-ip.net) |
| 16:08.55 | jrad | Morning, I have found myself with a strange issue. I have two asterisk boxs that connect to each other through VPN internally. It's worked for over a year now and today they are 'unreachable' to each other. Even though I can ping and interact with each box through the network. Any ideas? I have tried everything including rebooting the devices in the middle of the day. |
| 16:10.02 | Greenlight | Any firewalls in place? Changes of IP addresses ? |
| 16:10.21 | jrad | two firewalls on both sides of the VPN tunnel. No changes of IP addresses. |
| 16:17.23 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.15.174) |
| 16:24.45 | Katty | looks in |
| 16:25.15 | *** join/#asterisk retentiveboy (~retentive@72.54.144.26) |
| 16:37.20 | *** join/#asterisk shadar (~eugene@37.113.202.81) |
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| 16:52.39 | *** part/#asterisk Elleni (3ec00582@gateway/web/freenode/ip.62.192.5.130) |
| 16:56.36 | *** join/#asterisk ghost75 (~trechber@dslb-092-075-061-251.pools.arcor-ip.net) |
| 16:59.54 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 17:04.55 | [TK]D-Fender | <PROTECTED> |
| 17:05.58 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 17:11.31 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 17:14.28 | *** part/#asterisk orn (~orn@2a01:8280:101:0:b922:7a52:599a:7b96) |
| 17:16.08 | *** join/#asterisk mattsl (~user@68.169.165.190) |
| 17:16.28 | jrad | Ok. Fender I will take a look at the DEBUG messages from the register attempts. |
| 17:17.20 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 17:19.07 | *** join/#asterisk shadar (~eugene@37.113.202.81) |
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| 17:44.21 | *** join/#asterisk phonebuff (~chatzilla@23.31.148.65) |
| 17:45.52 | phonebuff | Good Afternoon -- Need some help with Codec Translations.. Anyone here that might be able to help ? |
| 17:48.03 | phonebuff | Specifically, 1.11.2 -- [2013-01-24 12:37:45] WARNING[10900][C-00000006] channel.c: No path to translate from SIP/2101-0000000a to SIP/2199-0000000b |
| 17:49.22 | phonebuff | Where 2101 is Codecs : (g729) Codec Order : (g729:20) |
| 17:49.23 | phonebuff | and 2199 is Codecs : (gsm|ulaw|alaw|h263) Codec Order : (ulaw:20,alaw:20,gsm:20,h263:0) |
| 17:53.52 | [TK]D-Fender | phonebuff, And did you install the licenses for your G.729? |
| 17:54.58 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 17:55.57 | phonebuff | You know I just realized that -- |
| 17:56.38 | phonebuff | Changed the 2101 from 729 and yep it works, now just attempting to get BLF indications.. |
| 17:57.16 | phonebuff | The lIcense is on the current 1.4 production machine that I am hoping to retire |
| 17:57.23 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 17:58.25 | [TK]D-Fender | phonebuff, Well there was your problem |
| 17:58.50 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:59.17 | phonebuff | Yes - One of many. Still fighting with getting BLF hints to the Cisco 7916 side bar on a Cisco 7975G |
| 18:00.23 | Mon|A|rch | if i want to record calls, is it better to use monitor or record? |
| 18:00.29 | Mon|A|rch | I'm a little fuzzy on the difference |
| 18:00.53 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 18:01.27 | ChannelZ | Monitor |
| 18:02.16 | ChannelZ | Record is for making a recording.. like to record your own IVR messages and things. |
| 18:02.25 | Mon|A|rch | okay |
| 18:02.29 | Mon|A|rch | cool, thanks |
| 18:02.42 | Mon|A|rch | what format does it output to? |
| 18:03.00 | ChannelZ | Monitor/MixMonitor you can think of as more of an asyncronous function, it records all the audio passing on the channel |
| 18:03.33 | Mon|A|rch | okay |
| 18:03.34 | ChannelZ | wav if you don't specify. see "core show application monitor" |
| 18:03.49 | Mon|A|rch | sweet |
| 18:03.50 | Mon|A|rch | thanks |
| 18:07.13 | Mon|A|rch | does it support mp3 natively for output? does it use SoX? |
| 18:11.36 | [TK]D-Fender | Mon|A|rch, It records in and format * can transcode, for which MP3 is not one. You can convert it in post-processing : "core show application monitor" |
| 18:14.01 | Mon|A|rch | [TK]D-Fender, so monitor(ogg,${RECORDING_NAME},m) would give me an ogg file? |
| 18:14.56 | [TK]D-Fender | Mon|A|rch, ogg is not a codec * supports... |
| 18:18.53 | Nugget | [TK]D-Fender: why do you hate Freedom? :) |
| 18:19.28 | Mon|A|rch | [TK]D-Fender, are there any lossy formats that it supports? |
| 18:19.30 | Mon|A|rch | ulaw? |
| 18:19.42 | Mon|A|rch | i just don't want to store all of these in .wav's |
| 18:20.47 | *** join/#asterisk elico (~Thunderbi@109.64.164.192) |
| 18:22.16 | din3sh | hello room, cat /proc/dahdi/1 shows Timing slips: 1 |
| 18:22.27 | din3sh | is this a problem? |
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| 18:24.37 | *** join/#asterisk devand (~devand@out-ab-220.wireless.telus.com) |
| 18:24.43 | devand | hey there. |
| 18:24.55 | devand | my asterisk keeps crashing, and i was wondering if someone more knowledgeable than myself might be able to take a look at my crashdump files to see if they can figure out why. |
| 18:25.08 | devand | i'm running asterisk 1.8 with Thirdlane PBX Manager (multi-tenant) as a front-end. running on RHEL 6.1 |
| 18:25.18 | devand | the problem only started occurring very recently, and is becoming more frequent, with 2 occurrences in the past 48 hours. |
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| 18:35.32 | cheasee | how do i find out which regional settings i have to set (on a pap2 ata) so things like call waiting and switching between both calls work? only information on this i could find: http://forum.voxilla.com/threads/pap2t-call-waiting-and-gigaset-r-key.32395/ but this seems to work for italy, how do i find out for austria? :) |
| 18:38.10 | Katty | lot of bluebirds on critter cam lately |
| 18:38.12 | Katty | infobot: crittercam |
| 18:38.12 | infobot | somebody said crittercam was Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
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| 18:44.55 | [TK]D-Fender | <[TK]D-Fender> Mon|A|rch, It records in and format * can transcode, for which MP3 is not one. You can convert it in post-processing : "core show application monitor" |
| 18:45.35 | [TK]D-Fender | cheasee, call-waiting and switching between calls isn't a "regional" thing. |
| 18:45.47 | Mon|A|rch | okay |
| 18:45.52 | [TK]D-Fender | cheasee, There is nothing to set beyond saying "yes, allow this feature" |
| 18:47.27 | Mon|A|rch | [TK]D-Fender, okay, i get it |
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| 18:57.42 | devand | Would anyone in here be able to take a look at a crash dump for me? I have an Asterisk server that is occasionally crashing on me, and I need to figure out why it's happening / how to fix it. Would be very much appreciated. |
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| 19:00.45 | cheasee | [TK]D-Fender: oh ok, well that siemens phone has a "R" button and according to the manual it should switch with that, but when i press the button it doesnt switch, instead the active caller hears a beep and i see the "R" on the display |
| 19:01.37 | cheasee | i googled for that and found that italian guy fixed this by changing that flash time setting, i tried that too but it didnt work, so i thought its a regional thing |
| 19:02.26 | cheasee | or is it maybe a analog phone thing and i have to find out the correct setting for my specific model? |
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| 19:11.21 | greenwolf | hello all... |
| 19:14.21 | Katty | hi |
| 19:14.45 | greenwolf | sup katty? |
| 19:14.53 | Katty | ceiling |
| 19:19.58 | cheasee | ah, i found the correct hook timer setting for pap2: http://www.velocityreviews.com/forums/t393586-hook-flash-delay.html Min: 0.06 Max: 0.2 works with gigaset now :) |
| 19:20.41 | greenwolf | something weird happen to my asterisk server |
| 19:20.53 | greenwolf | the audio stopped working for internal extensions on my LAN network |
| 19:21.04 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 19:21.12 | greenwolf | but when external calls in from my sip trunks or from outside lines...the audio works perfectly fine |
| 19:21.29 | greenwolf | i checked all my settings in the sip.conf files and per extensions also |
| 19:22.19 | greenwolf | everything seems fine and no problems but yet no audio is coming thru and NAT isn't the problem because its on LAN internal network, if there were a problem due to nat it wouldn;t work for outside trunks coming in |
| 19:22.37 | cheasee | greenwolf: how "the audio stopped working" ? |
| 19:22.59 | *** part/#asterisk dr0ck (~dr0ck@c-67-172-153-201.hsd1.co.comcast.net) |
| 19:23.46 | cheasee | in both directions or just one? |
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| 19:24.37 | greenwolf | well when i enter an IVR menu dialing from an internal extension no sound is emiited but yet asterisk continues dialplan like theres no problem |
| 19:24.47 | greenwolf | it shows its playing the menu and waiting waitexten() |
| 19:24.58 | greenwolf | but no audio on the softphone on the internal LAN |
| 19:25.16 | greenwolf | but if i call that same IVR menu using IPkall number from outside the system works fine = with audio |
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| 19:27.01 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:28.57 | cheasee | you are sure the volume of your phone isnt muted? ;) |
| 19:30.39 | *** join/#asterisk devand (~devand@out-ab-240.wireless.telus.com) |
| 19:31.18 | cheasee | can you call another internal extension or is it the same like calling the ivr? |
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| 20:49.05 | Mon|A|rch | I'm getting an unusual error |
| 20:49.18 | Mon|A|rch | I'm not sure if this is related to asterisk, or to the SIP provider we're using |
| 20:49.47 | Mon|A|rch | Got SIP response 500, and Got SIP response 404 are popping up |
| 20:50.03 | Mon|A|rch | saying address incomplete and internal server error? |
| 20:50.40 | *** join/#asterisk tmkt (~david@206.223.170.189) |
| 20:50.49 | tmkt | rejected because extension not found in context 'polycom'. |
| 20:50.51 | tmkt | any idea how to fix that? |
| 20:51.00 | tmkt | the phone can receive incoming calls at its extension |
| 20:51.06 | tmkt | but fast busy signal on the way out |
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| 21:00.27 | tmkt | ok..so now i have the phone working it can call an extension and receive calls from an extension..but can't seem to call out |
| 21:07.49 | [TK]D-Fender | tmkt, "sip set debug on" <- look at what it's looking for and where.... because whatever and wherever that is .. is not where it is looking |
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| 21:11.37 | danfromuk | Hi. Using sip.conf realtime, issueing a 'sip reload' clears all registrations. Does the same happen when using standard sip.conf files without realtime? |
| 21:13.48 | tmkt | found it |
| 21:13.52 | tmkt | all good..thanks. |
| 21:14.27 | tmkt | is unistim flaky? |
| 21:14.41 | tmkt | debating replacing all these nrotel phones with polycom sip phones |
| 21:14.59 | tmkt | trixbox crashes randomly when during phone calls |
| 21:15.14 | tmkt | and the previous operations person suggested it was due to the unistim phones |
| 21:15.58 | [TK]D-Fender | tmkt, UniSTIM is bastard proprietary protocol. The faster you dump it the better |
| 21:16.04 | [TK]D-Fender | Polycom > All |
| 21:18.22 | tmkt | ok |
| 21:18.23 | tmkt | will do |
| 21:18.27 | tmkt | last question |
| 21:18.38 | tmkt | Just transfered a phone from an old employee to a new one |
| 21:18.50 | tmkt | but the calls are stil comming from the old employees Call ID |
| 21:19.07 | tmkt | is that something I fix in trixbox? or at the provider? (Unlimitel) |
| 21:20.52 | drmessano | ..... |
| 21:22.47 | [TK]D-Fender | tmkt, Your phones are your phones..... that is for you to set up |
| 21:23.02 | tmkt | ok..so in Trixbox somewhere i'm missing a setting |
| 21:23.19 | tmkt | i'll see what I can track down |
| 21:23.20 | tmkt | thanks |
| 21:31.35 | tmkt | doh |
| 21:31.43 | tmkt | it was my iphone contact list |
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| 21:48.36 | phix | [TK]D-Fender: Polycom > Snom? |
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| 22:14.15 | citrusfizz | is g722 enabled by default in asterisknow? |
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| 22:18.31 | lvlinux | hi |
| 22:18.44 | lvlinux | I have a question: |
| 22:19.12 | lvlinux | i have an asterisk server set up with a sip trunk to another server with username "ggasterisk" |
| 22:19.37 | lvlinux | i have a phone connected to my asterisk server with extension/username "phone1" |
| 22:20.01 | lvlinux | my asterisk server registers to the other server fine |
| 22:20.32 | lvlinux | when i try to call an extension at the remote server over the sip trunk with phone1, the other server gives my server a 401 unauth |
| 22:21.44 | lvlinux | looking at the headers of the SIP messages, it shows my connection request from my asterisk server as being From: "phone1" <sip:ggasterisk@myipaddress> |
| 22:22.12 | lvlinux | any ideas why I'm getting the 401? |
| 22:23.13 | lvlinux | or rather any ideas why my asterisk server is saying the SIP invite is from phone1 instead of from itself? |
| 22:28.23 | [TK]D-Fender | phix, yup |
| 22:34.29 | lvlinux | it's actually YATE on the other end BTW |
| 22:37.13 | lvlinux | anyone have an idea where I should look for help? |
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| 23:54.35 | Mon|A|rch | hey, I'm getting the error: 484 "incomplete address" returned from my SIP provider |
| 23:54.50 | Mon|A|rch | is that an asterisk response? or a provider-specific response |
| 23:57.25 | Mon|A|rch | i guess what would be more helpful, is which log file has a log of the ami commands received? |
| 23:57.26 | Mon|A|rch | is there one? |
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