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| 01:49.46 | epaphus | hello all. Sorry iam a newbiw. I have an asterisk installation up.. how do i monitor in realtime the activity of asterisk? |
| 01:50.49 | WIMPy | rasterisk |
| 01:57.39 | epaphus | WIMPy, thanks... it says i have verbosity at least 3... how can I increase that to the maximum? |
| 02:00.28 | WIMPy | core set verbose ... |
| 02:06.21 | epaphus | This is interesting,iam running Verbosity to 15... and i thought that when dialing in from my DID i would see the caller ID .. but even at that verbosity it doesnt show |
| 02:06.23 | epaphus | hmmm |
| 02:09.07 | WIMPy | That's what we have Verbose and NoOp for. |
| 02:11.00 | epaphus | WIMPy, can you please provide me example.. so i can see what i need? |
| 02:11.23 | WIMPy | Write your dialplan. |
| 02:11.30 | [TK]D-Fender | epaphus: "core show application NoOp" |
| 02:11.40 | WIMPy | And start by reading the |
| 02:11.47 | WIMPy | ~book |
| 02:11.47 | 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 |
| 02:12.39 | epaphus | thanks. I already do have a dial plan. Iam just intending to see the caller ID of the party rining the DID.. thats all.. from CLI |
| 02:14.32 | [TK]D-Fender | "core show function CALLERID" |
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| 02:43.05 | Platinum | Hello |
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| 02:49.48 | Platinum | hello |
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| 04:40.24 | danfromuk | Hi. Out of interest, does it add a great deal to the server load if I add different music on hold music for each incoming DID? 30 calls on hold at any time. |
| 04:40.51 | danfromuk | All 30 calls listening to different music on hold |
| 04:41.03 | danfromuk | Compared with having one default music on hold that all calls hear. |
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| 05:03.14 | dgeary2 | i have compiled asterisk with chan_sip, res_crypto and res_srtp. is there anything else needed to support SIP/TLS with SRTP with SDES? |
| 05:03.35 | *** join/#asterisk jkroon (~jkroon@dsl-244-23-142.telkomadsl.co.za) |
| 05:03.45 | dgeary2 | the TLS connection is being dropped after the client hello but before the server transfers the server certificate |
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| 05:06.48 | dgeary2 | i'm using asterisk-1.8.16.0 |
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| 05:30.10 | jkroon | would someone be able to help me troubleshoot why my E1 link stays Down, Active? L1 according to dahdi_scan seems to be OK |
| 05:31.09 | jkroon | I've even looped two ports on the T410P card using a cross to see if that comes up, but both those ports stays down, active too, thus I'm reasonably sure it has to be a config issue ... |
| 05:32.14 | jkroon | switchtype is euroisdn |
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| 05:34.14 | jkroon | signalling is pri_cpe (except for port 1 which is pri_net), port 1 links to port 2, port 4 to the telco |
| 05:45.05 | jkroon | TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) <-- any ideas/suggestions? |
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| 06:28.37 | jkroon | tzafrir_laptop, you around? |
| 06:28.50 | tzafrir_laptop | yes, for a short while |
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| 06:29.34 | jkroon | thanks, i've got a te205p card i'm having some issues with, remote access to the machine only, so hard to log a call with digium, do you have a few minutes to help? |
| 06:29.48 | jkroon | (can't get the serial) |
| 06:30.24 | jkroon | basically /proc/dahdi/* reports the channels being OK (dahdi_scan), but in asterisk pri show spans leaves then "Down, Active", I've no idea where to start trouble-shooting. |
| 06:31.00 | jkroon | SABME messages seems to be sent, but no responses received. libpri 1.4.12, dahdi{,-tools} 2.6.1 and ast 1.8.15.1. |
| 06:33.31 | jkroon | http://pastebin.com/NAe6hhVv <-- system.conf and dahdi config in asterisk. |
| 06:34.18 | jkroon | ports 1 and 2 is looped using a x-over cable (thus why setting pri_net on port 1) |
| 06:34.19 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 06:34.50 | jkroon | Communication controller: Digium, Inc. Wildcard TE405P/TE407P quad-span T1/E1/J1 card 5.0V (rev 02) <-- lspci |
| 06:35.40 | tzafrir_laptop | No alarms? |
| 06:36.32 | jkroon | none |
| 06:36.43 | jkroon | alarms all report OK |
| 06:37.52 | jkroon | http://pastebin.com/QyRiRxcp <-- dahdi_scan |
| 06:38.21 | jkroon | ok, span 3 is RED/LFA but it's not connected either |
| 06:39.34 | tzafrir_laptop | sorry, no idea |
| 06:39.39 | tzafrir_laptop | GTG |
| 06:39.44 | jkroon | thanks for looking |
| 06:39.53 | jkroon | will try to get the serial number and report to digium. |
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| 07:24.42 | schmidts | good morning |
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| 07:35.28 | beebeeep | в отличие от квалифицированных человекочасов |
| 07:35.38 | beebeeep | sorry |
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| 07:38.12 | ChannelZ | In soviet Russia, keyboard types you! |
| 07:41.10 | schmidts | ChannelZ +1 ;) |
| 07:45.05 | ChannelZ | infobot: karma for ChannelZ |
| 07:45.05 | infobot | for channelz has neutral karma |
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| 07:57.55 | jkroon | ok ... that's a first ... required a full power cycle to fix PRI problem. |
| 07:58.02 | jkroon | never seen that one before. |
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| 08:08.07 | gusto | why should keyboard type something? |
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| 10:00.26 | fling | how to record all my calls? |
| 10:01.14 | bulkorok | <PROTECTED> |
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| 10:31.56 | tompaw | Hello! |
| 10:37.58 | fling | bulkorok: thanks |
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| 11:20.36 | mtbf | Which unix command is used by Monitor on asterisk 1.8? I thought it was sox, but I get recordings even when I move the binary. |
| 11:25.15 | ectospasm | sox is only used when you pass the right option to Monitor |
| 11:25.21 | tompaw | ~pastebin |
| 11:25.21 | 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. |
| 11:25.22 | ectospasm | MixMonitor mixes the files on the fly |
| 11:27.09 | mtbf | ectospasm: I just changed the dialplan to avoid using it, since I want to preserve recordings for separated channels too, I have ready overlay in bash for my copy of sox, which copies $1 and $2 to dedicated directory, then runs sox -m $1 $2 $3. I use Monitor(wav,filenamewithoutext,m) in my dialplan for that. What should I change? |
| 11:28.15 | ectospasm | well, you should get recordings... one labeled -in, the other labeled -out |
| 11:30.15 | mtbf | Yes, I am getting them, and I want Monitor to run sox (my overlay actually) to mix them. |
| 11:30.25 | tompaw | Hello, I've been using an Agent Queue system with the ability to add 3rd parties and I'm looking to optimize it. The dialplan is available here: http://pastebin.ca/2207143 |
| 11:30.50 | ectospasm | mtbf: so why move the sox binary then? |
| 11:30.55 | tompaw | It was built on 1.8 and now I'd like to migrate to 10. |
| 11:31.15 | ectospasm | tompaw: don't migrate to 10 unless it has features you really need. Wait for 11 to be released. |
| 11:31.22 | mtbf | ectospasm: I don't want the -in and -out files to be removed after mixing. |
| 11:31.52 | tompaw | ectospasm: any reason why? last time I mentioned MeetMe and ConfBridge here, everyone was like MIGRATE TO X NAO!!! |
| 11:32.39 | ectospasm | mtbf: if you want to save the two separate files, you'll have to mix them separately (outside of Asterisk) |
| 11:33.09 | ectospasm | tompaw: 10 is only a standard release. It will go into security-fix only mode in December |
| 11:33.31 | ectospasm | 1.8, and 11 are both LTS (Long Term Support) releases |
| 11:33.37 | tompaw | Also, since I'm planning to use this part of code commercially, I wouldn't mind if someone offered me a paid support (providing you can invoice me). |
| 11:33.47 | tompaw | ectospasm: understood. |
| 11:34.27 | ectospasm | tompaw: http://www1.digium.com/en/products/asterisk/support |
| 11:34.40 | ectospasm | disclaimer: I work for Digium Technical Support |
| 11:34.53 | tompaw | Nah, I'd rather deal with someone from here directly. I don't have time nor nerves to deal with enterprise support systems. |
| 11:35.07 | ectospasm | suit yourself. |
| 11:35.42 | tompaw | The thing is - it worked fine, but I simply do not feel 100% convinced if the idea is correct, especially the cont-agents-conf context. |
| 11:36.00 | ectospasm | mtbf: oops, you can set ${MONITOR_EXEC}, and it'll keep the files. Sorry |
| 11:36.45 | ectospasm | tompaw: if it works, how can it be wrong? |
| 11:37.31 | tompaw | ectospasm: I just got a bad feeling about it. I'd rather someone helped me confirm the actuall call leg flows through the dialplan. |
| 11:37.59 | tompaw | And yeah, I wrote it myself, but I only got it to a stage where it works, without full understanding of the process. |
| 11:38.31 | ectospasm | ...and it's too late for me to be troubleshooting dialplan |
| 11:40.32 | tompaw | does digium to per-incident support without signing the whole year agreement? |
| 11:41.12 | ectospasm | you get two incidents at the lowest level... you get a year to use them up |
| 11:41.14 | tompaw | s/to/do/ |
| 11:41.49 | tompaw | ectospasm: as a Digium expert, can you please let me know if what I'm after qualifes as a single incident? |
| 11:42.23 | ectospasm | any question that involves a fair amount of analysis and research will qualify for an incident |
| 11:42.42 | ectospasm | your issue would qualify |
| 11:42.50 | tompaw | Thanks. |
| 11:42.52 | ectospasm | ...I haven't looked at your dialplan yet |
| 11:43.11 | ectospasm | ...but in order to look at the dialplan we'd need to incur an incident |
| 11:43.14 | tompaw | it's like 30+ lines including comments and 6 different context. |
| 11:43.18 | ectospasm | ...someone here may do it for cheaper. |
| 11:43.36 | ectospasm | yeah...it could be two lines of dialplan, it wouldn't matter |
| 11:43.44 | ectospasm | one thing we will not do is write it for you |
| 11:43.51 | ectospasm | ...but since you've already done that |
| 11:44.15 | tompaw | do you do the independant consultations or only via digium? |
| 11:44.48 | ectospasm | I would be competing with my employer if I did them independently (; |
| 11:45.09 | tompaw | fair enough |
| 11:45.49 | ectospasm | anyway, I need to get to bed. |
| 11:45.57 | tompaw | good night. |
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| 11:54.38 | *** join/#asterisk oquidave (~oquidave@41.223.84.34) |
| 11:54.43 | oquidave | hello |
| 11:55.18 | oquidave | i've just encountered something rather odd in asterisk scripting(perhaps am new)... |
| 11:55.28 | oquidave | and it's do with expressions...same => n,GotoIf($[${TEST} = 1]?weasels:iguanas) |
| 11:56.11 | oquidave | so in asterisk "=" operator is both an equality tester operator as well as an assignment operator? |
| 11:56.32 | oquidave | as opposed to "==" vs "=" ?? |
| 11:57.28 | tompaw | oquidave: to be honest, I was always assignint values using Set() |
| 11:58.19 | oquidave | tompaw, correct and i've no problem with that as in exten => 345,1,Set(TEST=1), but... |
| 11:58.49 | tompaw | what I'm saying is, it's not the '=' char that's doing the assignment, but the Set function. |
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| 11:59.15 | oquidave | to use the same operator for equality testing is really odd according to my experience. it's access a possible cause of confusion |
| 11:59.39 | oquidave | tompaw> okay so what do u use to test equality? |
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| 11:59.49 | jaxon007_ | how to make rpm for dahdi from source? |
| 12:03.05 | tompaw | oquidave: you just answered that yourself... Gotoif? |
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| 12:04.20 | oquidave | tompaw okay,though it's rather odd coming from standard scripting languages...but am fine with it |
| 12:04.58 | oquidave | tompaw i've been up until now using alpahnumeric names for my device names in the sip,conf, such that in the dialplan in the extensions.conf file i can write exten => 20,1,Dial(SIP/oquidroid,30); |
| 12:05.28 | oquidave | however, if am dealing with 1000+ clients/users, this methods seems to be inefficient |
| 12:06.10 | tompaw | Did you consider using external registry sources, like an sql database? |
| 12:06.21 | oquidave | as i have to manually specify the device names in the sip.conf file and then appropriately write it the extensions.conf file SIP/devicename |
| 12:06.59 | tompaw | I am using res_pgsql + extconfig (sipuser + sippeers) to work around static sip.conf files. |
| 12:07.07 | tompaw | You can then integrate it nicely with your system. |
| 12:08.07 | oquidave | so, am thinking that i need to use pattern matching style instead in the dialplan instead...like exten=>_NX, 1, Dial(SIP/${EXTEN}) |
| 12:08.58 | oquidave | tompaw oh....that's new |
| 12:09.24 | oquidave | tompaw oh....haven't looked at that...have a link to a tutorial? |
| 12:10.08 | tompaw | Nope, but using the keywords I gave you, it should be quite easy to find it yourself. |
| 12:10.27 | tompaw | Bear in mind, I am no a Digium representative and it's not neccessarily the best option for you. It just works for me. |
| 12:12.39 | oquidave | tompaw> but in a nutshell, you are storing users details in a db(mysql for instance) and then some script writes the details in a sip.conf file? is that it? |
| 12:13.04 | tompaw | nope |
| 12:13.07 | *** join/#asterisk italorossi (~textual@189.124.194.15) |
| 12:13.18 | tompaw | I am storing those details in the sql, and then asterisk queries that sql each time it needs. |
| 12:14.15 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:14.20 | tompaw | So if someone's trying to register, Asterisk fires an SQL query to the DB and gets all the equivalents of the static .conf values. Afterwards it simply goes on with the registration. |
| 12:16.08 | oquidave | tompaw> sounds alittle abstract for now since all i know upto this point is static sip.conf and extension.conf file that manually have to be configured...anyway i'll google it up |
| 12:16.55 | tompaw | I cannot imagine handling a sip.conf file with thousands of entries to be honest... |
| 12:17.58 | tompaw | oquidave: as I recall, I had a big problem building the tables to meet asterisk's criteria, since it wasn't documented anywahere. I ended up sniffing the queries and simply matching what * needs in my db structure. |
| 12:18.13 | tompaw | When you get to that stage, I don't mind sharing the table structure dump with you (pgsql). |
| 12:19.17 | *** join/#asterisk Joel_re (~jr@115.114.59.182) |
| 12:19.19 | Joel_re | hey |
| 12:19.22 | wdoekes | tompaw: ./contrib/realtime/mysql/sippeers.sql and files in the vicinity |
| 12:19.35 | Joel_re | was wondering if those voip weekend chats are still in effect |
| 12:19.42 | Joel_re | on friday evenings |
| 12:19.52 | Joel_re | sorry to be offtopic :\ |
| 12:20.13 | tompaw | wdoekes: thanks, I wish I knew it a year ago :) |
| 12:20.39 | wdoekes | haha.. I didn't know about those files either. I checked the source instead |
| 12:20.56 | oquidave | tompaw that db does asterisk use for it's backend? |
| 12:21.05 | tompaw | yep |
| 12:21.35 | tompaw | wdoekes: I wonder how long they've been there. I remember a year ago I came across a similar .sql, but it was VERY different from what my asterisk expected. |
| 12:21.59 | wdoekes | they get updated at irregular intervals ;) |
| 12:26.43 | *** join/#asterisk deo_ (~dnepangue@203.177.214.75) |
| 12:28.00 | *** part/#asterisk deo_ (~dnepangue@203.177.214.75) |
| 12:34.16 | oquidave | am having a little bit of challenging registering devices with numeric names...i get ... chan_sip.c:25160 handle_request_register: Registration from '"oquidave"<sip:oquidroid@192.168.43.240;transport=UDP>' failed for '192.168.43.1:5060' - No matching peer found |
| 12:35.04 | oquidave | oquidroid is a sip client with username=oquidroid but the device name is 20 as specified in the sip.conf file like [20] |
| 12:35.29 | oquidave | why would that be a problem? |
| 12:35.30 | *** join/#asterisk biomorph (~m@94.116.180.154) |
| 12:37.13 | oquidave | when i sip show peers, i've 20/oquidroid<20> (Unspecified) D N 0 Unmonitored |
| 12:37.23 | *** join/#asterisk biomorph (~m@94.116.180.154) |
| 12:39.17 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:39.17 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:39.49 | leifmadsen | Joel_re: are you talking about the VUC? /join #vuc -- there are talks every Friday at 1200 EDT (-0400 GMT) |
| 12:41.42 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 12:42.10 | *** join/#asterisk biomorph (~m@94.116.180.154) |
| 12:43.51 | oquidave | quick qn: does the device/client username have to the same as the device name in the sip.conf file? |
| 12:44.28 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:45.09 | wdoekes | oquidave: username= does not do what you expect. you want [username] |
| 12:45.45 | Joel_re | leifmadsen: thanks! |
| 12:46.02 | *** part/#asterisk jaxon007_ (~jay@123.252.144.92) |
| 12:46.52 | oquidave | wdoekes> yes but i just googled around and found that the device name specied in [devicename] as numeric is the same as username=devicename...mine are different, so i thought that could a problem |
| 12:52.48 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 12:57.33 | jeffspeff | I'm trying to set a phone/sip user to have a static host/ip instead of dynamic. Asterisk is giving an error that the Peer is not supposed to register. Is this because of the "regexten" setting for the sip user or because I have a secret set? |
| 13:00.36 | carrar | You still need to configure asterisk as if the phone was still dynamic |
| 13:00.41 | carrar | if you want it to register |
| 13:00.47 | *** join/#asterisk serafie (~erin@nat/digium/x-jwvowwshngynfjmo) |
| 13:01.19 | jeffspeff | i did, but asterisk is giving that error and when that phone receives a call, the CID info is all messed up and shows a sip string |
| 13:01.53 | carrar | You shouldn't have made any changes to Asterisk when you made the phone a static IP |
| 13:02.21 | jeffspeff | the phone was working fine, registering properly, etc as a dynamic host. I changed the "host" value from "dynamic" to the users external IP. I didn't change anyhting else |
| 13:02.41 | carrar | put it back to dynamic |
| 13:02.47 | carrar | which is what I just said |
| 13:02.53 | jeffspeff | uhm that's not what i'm trying to accomplish |
| 13:02.58 | carrar | yes it is |
| 13:03.28 | jeffspeff | carrar, if i want the phone to register from only a specific IP that specify then host needs to be something other than "dynamic" |
| 13:03.39 | carrar | no |
| 13:03.50 | carrar | otherwise it's not registering |
| 13:04.14 | carrar | Asterisk doesn't care if the phone is dhcp or static |
| 13:04.23 | jeffspeff | that's fine, but I do |
| 13:04.30 | carrar | if you still want to register, you need to make that dynamic |
| 13:05.11 | jeffspeff | ok, so back to my original question. Is it trying to register because of the "regexten" setting or because I have specified a "secret" ? |
| 13:06.16 | carrar | because the phone is configured to register |
| 13:07.07 | *** join/#asterisk serafie1 (~erin@75.76.38.159) |
| 13:07.34 | jeffspeff | let's try this again |
| 13:07.51 | jeffspeff | what in the sip configuration tells the phone to register? |
| 13:08.03 | carrar | nothing |
| 13:08.08 | carrar | the phone does |
| 13:08.13 | jeffspeff | If I remove the secret from sip and the phone, will it still register? |
| 13:08.44 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-hiyawymnboatfiyk) |
| 13:08.44 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:09.31 | schmidts | jeffspeff if you remove the secret row, instead of setting a new secret then this peer can register even without a password |
| 13:09.58 | carrar | registering is so the server knows where to reach the phone |
| 13:10.42 | *** join/#asterisk Katty (~Katty@96-35-244-101.dhcp.stls.mo.charter.com) |
| 13:10.43 | jeffspeff | Ok, so i want to specify a static host for each of my phones. I want the registration error to go away. how do i accomplish this? |
| 13:11.06 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 13:11.20 | [TK]D-Fender | jeffspeff, If you tell * that it has a fixed host then it is not ALLOWED to register |
| 13:11.24 | [TK]D-Fender | jeffspeff, Those are the rules. |
| 13:11.37 | [TK]D-Fender | jeffspeff, Tell your phone to stop registering |
| 13:11.43 | carrar | yes |
| 13:11.59 | jeffspeff | i don't recall seeing a register on/off toggle |
| 13:12.13 | oquidave | hello guys, here's my sample sip.conf and extension.conf http://pastebin.com/ZDgB72G9 i don't understand why the device named "20" is not registering!! |
| 13:13.18 | oquidave | previously the device name was alphabetic and it it worked fine, now am experimenting with numeric device names...but so far the device aren't registering |
| 13:13.41 | [TK]D-Fender | oquidave, First remove the comment from the end of the device entry. Next taht is only a small segment of a proper sip.conf. next we have no proof that any registration attempt is even reaching your server and what , if anything, asterisk is responding to it. |
| 13:14.25 | [TK]D-Fender | oquidave, in other words : Don't show us a brochure picture of your car and ask us why the brakes failed |
| 13:14.26 | oquidave | <PROTECTED> |
| 13:14.57 | tompaw | [TK]D-Fender: with some cars, the brochure is all you need. |
| 13:15.10 | [TK]D-Fender | oquidave, username = oquidroid;call extension 20 to get to the droid <- stop shoving COMMENTS right at the end with no spacing |
| 13:20.04 | carrar | jeffspeff, curious why you would not want your phones to register? |
| 13:20.15 | carrar | Just playing? |
| 13:20.26 | oquidave | [TK]D-Fender done that but i still get chan_sip.c:25160 handle_request_register: Registration from '"oquidave"<sip:oquidroid@192.168.43.240;transport=UDP>' failed for '192.168.43.1:5060' - No matching peer found |
| 13:21.19 | [TK]D-Fender | oquidave, I don't see updated configs, nor "sip show peers" to show that * even acknowledges any of your config |
| 13:21.32 | [TK]D-Fender | Maybe soon we'll hit "thorough" |
| 13:21.44 | oquidave | here 20/oquidroid (Unspecified) D N 0 Unmonitored |
| 13:22.31 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:23.03 | *** join/#asterisk mKn0wt (~mtv@208.96.137.16) |
| 13:23.31 | [TK]D-Fender | oquidave, Trash all the comments and actually set the type to peer. Then also show us "sip show peer 20" and the full registration attempt..... |
| 13:24.59 | oquidave | http://pastebin.com/9arTyGEz |
| 13:29.02 | *** join/#asterisk scgm11 (~Sebastian@r186-50-67-112.dialup.adsl.anteldata.net.uy) |
| 13:30.10 | scgm11 | hi, is there any way of doing macros async??? I would like to make a call and have a macro executing while the call is up, and depending of something that happens in the macro the call could be hanged, etc. |
| 13:30.14 | *** join/#asterisk bchia (~Adium@nat/digium/x-diqnrhroypuosjgp) |
| 13:30.59 | oquidave | [TK]D-Fender> here http://pastebin.com/BvCbJu8q thanks |
| 13:31.28 | oquidave | sory i set the type=peer |
| 13:32.13 | [TK]D-Fender | sip et debug on <------------------------ |
| 13:32.15 | [TK]D-Fender | sip set debug on <------------------------ |
| 13:33.54 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:33.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:34.08 | oquidave | [TK]D-Fender> it's enabled |
| 13:34.09 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:34.38 | oquidave | [TK]D-Fender let me send the debug again |
| 13:34.56 | scgm11 | hi, is there any way of doing macros or gosub async??? I would like to make a call and have a macro executing while the call is up, and depending of something that happens in the macro the call could be hanged, etc. |
| 13:35.59 | oquidave | [TK]D-Fender but it's too long!! but let me send everything |
| 13:36.22 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:36.22 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:36.27 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 13:36.29 | [TK]D-Fender | scgm11, you will have to launch another channel. You can do this from a macro or gosub called by the dial upon answer. |
| 13:36.59 | scgm11 | that won't work, because is not async |
| 13:37.05 | scgm11 | when I pass the control to the macro |
| 13:37.08 | scgm11 | stays there |
| 13:37.10 | scgm11 | for example |
| 13:37.19 | scgm11 | make a dial with the macro |
| 13:37.31 | scgm11 | the macro has a while in there checking for something |
| 13:37.37 | scgm11 | until the macro is not over |
| 13:37.48 | scgm11 | if I answer the call |
| 13:37.51 | scgm11 | I have no audio |
| 13:38.07 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust1000.sgyl.cable.virginmedia.com) |
| 13:38.09 | oquidave | [TK]D-Fender here updated....http://pastebin.com/kAdBTAkm |
| 13:38.09 | scgm11 | because seems that the macro has the control |
| 13:38.51 | [TK]D-Fender | scgm11, I did NOT say to stay in there |
| 13:39.01 | [TK]D-Fender | scgm11, Read what I told you again. |
| 13:39.18 | scgm11 | create another channel from the macro? |
| 13:39.41 | [TK]D-Fender | Was I somehow not explicit enough there? |
| 13:39.58 | scgm11 | english is not my first language ;) |
| 13:40.12 | [TK]D-Fender | <[TK]D-Fender> scgm11, you will have to launch another channel. |
| 13:40.37 | [TK]D-Fender | Call file. Originate. Voodoo |
| 13:41.01 | scgm11 | ok thanks I will make some tries |
| 13:41.19 | IsUp | hiya. anyone able to place a test to my cellphone? i am trying to find a voip provider which has access to Turkey. |
| 13:42.29 | [TK]D-Fender | <--- Transmitting (NAT) to 192.168.43.1:5060 ---> |
| 13:42.29 | [TK]D-Fender | SIP/2.0 403 Forbidden (Bad auth) |
| 13:42.35 | [TK]D-Fender | I'm seeing straight up bad auth... |
| 13:43.02 | [TK]D-Fender | certainly not trusting your zoiper side... |
| 13:43.43 | Greenlight | Howdy all. Got a strange issue with Asterisk, that I'm unsure how to progress. On very busy server (500+ channels) every few weeks Confbridge Record via the AMI causes a deadlock. I get absolutely no further output over the AMI, until I log into console and execute a "confbrdige record stop". |
| 13:45.15 | oquidave | [TK]D-Fender> i've also seen that, but it was working when the device name was alphatic, all i did was to change the device name in the sip.conf file to numeric...i suppose the rest of the config on zoiper remain e.g host, username, password, |
| 13:46.05 | [TK]D-Fender | redo it, and then test another client |
| 13:48.25 | Greenlight | Any ideas how to debugg such an issue? I dont think I can enable the debug locks thing as the server is too busy |
| 13:48.57 | oquidave | oaky |
| 13:53.15 | oquidave | [TK]D-Fender i changed back to alphabetic names and the device is registered!!! without any problems |
| 13:53.47 | oquidave | [TK]D-Fender is this problem specific to zoiper only? |
| 13:54.00 | [TK]D-Fender | oquidave, switch back to [20] and change "username" to "defaultuser" |
| 13:54.29 | oquidave | okay |
| 13:57.26 | oquidave | [TK]D-Fender Nope, it doesn't work |
| 13:57.53 | [TK]D-Fender | hrm |
| 13:58.57 | wdoekes | oquidave: register your client as user "20" |
| 13:59.51 | tompaw | oquidave: in zopier, fill in the username as both the user name and the caller id |
| 13:59.55 | tompaw | that should help you |
| 14:00.03 | tompaw | I only now realized you're using zoiper. |
| 14:00.17 | tompaw | so enter "20" in both of those fields. |
| 14:02.15 | oquidave | tompaw but it's the device name which is 20 and username is oquidroid !! does just work magically? |
| 14:02.38 | tompaw | one sec. in Zoiper, you have two fields: "Username" and "Caller ID Name", right? |
| 14:02.44 | oquidave | correct |
| 14:02.57 | tompaw | enter 20 in both, leave [20], delete username=, try again. |
| 14:03.08 | oquidave | okay |
| 14:03.11 | *** join/#asterisk dms (~dms@nat/digium/x-tbktrljyvklmnvqw) |
| 14:03.47 | oquidave | meaning i also don't need to specify "secret" in sip.conf and neither in zoiper |
| 14:04.56 | tompaw | noooo |
| 14:04.59 | tompaw | leave secret as it is |
| 14:05.05 | tompaw | and OBVIOUSLY leave it in Zoiper |
| 14:05.20 | wdoekes | oquidave: 14:45 < wdoekes> oquidave: username= does not do what you expect. you want [username] |
| 14:06.16 | wdoekes | [username] and secret=SECRET are the relevant config items |
| 14:06.17 | oquidave | tompaw if i don't specify the password field, zoiper doesn't allow me to save the configs |
| 14:06.28 | tompaw | JEsus Christ, leave the password there! |
| 14:07.00 | tompaw | I'm only saying you should have 20 in both "Username" and "Caller ID Name" |
| 14:07.18 | wdoekes | "Name" sounds irrelevant |
| 14:07.31 | oquidave | tompaw> done that ...and it now WORKS |
| 14:07.43 | oquidave | but why or how it works is still a mystery |
| 14:07.47 | tompaw | I know why |
| 14:08.07 | tompaw | because for the last few years Zoiper and Asterisk couldn't come to terms on user authentication |
| 14:08.25 | tompaw | What Asterisk expects as a username must be in both those fields in Zoiper for it to work. |
| 14:09.12 | oquidave | tompaw> so you are saying this config is only unique to zoiper |
| 14:09.27 | tompaw | no, your config is typical: [username] and secret |
| 14:09.49 | tompaw | but what unique to zoiper is that if you're using numerical logins you need them in BOTH of these fields or else. |
| 14:10.36 | oquidave | tompaw, okay thanks...am sure anyother new guys, i going to face hell..because it's not anyway mentioned like that!! |
| 14:11.07 | tompaw | you're welcome, whatever you said afterwards. |
| 14:11.10 | *** join/#asterisk dms (~dms@nat/digium/x-dnnhpcohxhfytzdq) |
| 14:16.46 | *** join/#asterisk pguima (186734f2@gateway/web/freenode/ip.24.103.52.242) |
| 14:17.29 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 14:19.48 | Katty | BACON STRIPS. |
| 14:19.52 | Katty | that is all. |
| 14:21.31 | Joel_re | hi Katty |
| 14:21.39 | Joel_re | <- smooth_penguin btw |
| 14:21.54 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 14:22.00 | Joel_re | been a while since I joined this channel |
| 14:24.20 | [TK]D-Fender | Katty, THAT'S SMART |
| 14:24.38 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 14:25.09 | Katty | Joel_re: smoooooth operator!!! |
| 14:25.14 | pguima | Good Morning, I was wondering if anyone ever had a problem where the sound of call through IAX trunk, between 2 asterisk servers, goes away for about 2 seconds and it returns. I am currently running asterisk 1.6.23.13. |
| 14:25.17 | Katty | Joel_re: why did you change your /nick? |
| 14:25.42 | Joel_re | heh, well I work for some org that happens to use freenode and I dont want to spook them |
| 14:25.58 | Joel_re | Katty: are you still keeping the squirrels fat? |
| 14:26.36 | oquidave | so to zoiper a [devicename] in sip.conf is the username in the client and you don't specify the username in the sip.conf file otherwise you're screwed...what a twist!! |
| 14:26.37 | Katty | sadly no :< |
| 14:26.47 | Katty | Joel_re: i moved to an apt... |
| 14:26.57 | Katty | Joel_re: and i'm afraid the little critters can't quite reach my balcony |
| 14:27.08 | Joel_re | :< |
| 14:28.06 | Katty | yes indeed. |
| 14:28.07 | oquidave | but thanks [TK]D-Fender and tompaw> |
| 14:28.11 | Katty | i /am/ keeping the cardinals fat tho :> |
| 14:31.20 | Joel_re | \0/ |
| 14:32.23 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 14:34.08 | carrar | katty, make a critter ladder |
| 14:34.32 | carrar | cause glue and popsicle sticks is fun |
| 14:34.33 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:35.16 | Katty | carrar: while i had a nice big branch that i tied to my balcony |
| 14:35.29 | Katty | carrar: but then the landlord decided he was going to chop the tree down |
| 14:35.42 | Katty | carrar: or at least that big branch....he thought squirrels were getting on the roof |
| 14:35.46 | carrar | You should make some rop bridges |
| 14:35.48 | Katty | carrar: i was not pleased, but took a hint. |
| 14:35.51 | carrar | rope |
| 14:35.55 | Joel_re | :\ |
| 14:35.57 | Katty | or i should just move. |
| 14:36.03 | carrar | Move to the NW |
| 14:36.06 | Katty | which i plan on doing soon. |
| 14:36.20 | carrar | great see you here soon!! :) |
| 14:38.26 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 14:39.26 | carrar | did you just move? |
| 14:39.33 | carrar | looks like it |
| 14:39.36 | Katty | it's been about 2 years ago. |
| 14:39.55 | carrar | heh |
| 14:39.59 | Katty | so probably 3 years when i move again |
| 14:40.05 | Katty | it's a long story. |
| 14:40.05 | carrar | I was talking about your irc login ip changing :) |
| 14:40.14 | Katty | yes, that too. |
| 14:40.25 | carrar | LONG STORYS are always so full of DRAMA |
| 14:40.27 | Katty | box at home was being goofy. |
| 14:40.32 | Katty | so now into box at work. |
| 14:40.38 | Katty | carrar: yes. |
| 14:40.41 | Katty | carrar: yes it is. |
| 14:40.44 | carrar | heh |
| 14:41.09 | Katty | carrar: and it's unfortunate. |
| 14:41.17 | Katty | sadly we can't always avoid drama. |
| 14:41.28 | carrar | yes |
| 14:41.31 | carrar | sadly that is true |
| 14:42.37 | carrar | So |
| 14:42.56 | Katty | sew. |
| 14:43.05 | carrar | Where you moving too in 3 years |
| 14:43.18 | carrar | Asia? |
| 14:43.21 | Katty | dunno yet. that will be the result of a very long, ocd process. |
| 14:43.24 | Katty | oh no. |
| 14:43.27 | Katty | i'll be in the same city. |
| 14:43.33 | Katty | i've not moved outside of my city. |
| 14:43.53 | carrar | You should go beyong the gates |
| 14:43.56 | Katty | i agree. |
| 14:43.57 | carrar | beyond |
| 14:44.02 | Katty | but it's bad timing, with my mother having alzheimers and all. |
| 14:44.15 | carrar | no other family members? |
| 14:44.32 | Katty | perhaps in a decade or so. |
| 14:44.41 | Katty | but i could move to stl...that is an option |
| 14:44.58 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:46.13 | Katty | not close enough to help. |
| 14:46.17 | carrar | I can't even count the number of times I've moved |
| 14:46.40 | Katty | they're either on the west coast, or the east coast. |
| 14:47.20 | Katty | it's not pleasant. i hate moving. |
| 14:47.30 | carrar | Buy a truck |
| 14:47.43 | carrar | get ride of items :) |
| 14:47.44 | carrar | thats key |
| 14:47.47 | carrar | rid |
| 14:48.29 | Katty | yeah. |
| 14:48.31 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 14:48.41 | carrar | put everything you own in a room |
| 14:48.51 | carrar | and only take out when you have to use |
| 14:48.53 | Katty | i'm not attached to most of the things i have. |
| 14:48.58 | carrar | dump everything else :) |
| 14:52.28 | Nugget | attaches himself to Katty |
| 14:54.05 | carrar | Are you binding to a port? |
| 14:56.42 | Katty | :> |
| 14:59.28 | slav3_kitten | god i really need to get around to setting up my asterisk box |
| 14:59.44 | carrar | we've been trying to call you for months |
| 15:10.09 | Katty | a very heavy smoker keeps coming into my office to work on something. |
| 15:10.16 | Katty | at this rate i'm going to run out of febreeze |
| 15:12.20 | *** join/#asterisk robl^laptop (~robl^@pdpc/supporter/active/robl) |
| 15:16.07 | *** join/#asterisk Daneo (~Daneo@109.132.202.117) |
| 15:20.32 | n3hxs | Spray it into their pack of cigarrettes |
| 15:21.17 | tompaw | It's illegal to smoke at work or public places in EU. Isn't that the case in Murrica? |
| 15:23.35 | n3hxs | Yes |
| 15:24.01 | *** join/#asterisk pbxman (c335d9a4@gateway/web/freenode/ip.195.53.217.164) |
| 15:24.14 | n3hxs | but it doesn't stop the person who is emanating used smoke from their clothes. |
| 15:24.34 | tompaw | It's illegal to smell or be fat at work or public places in EU. Isn't that the case in Murrica? |
| 15:25.15 | n3hxs | LOL |
| 15:25.39 | Katty | tompaw: i don't think you can legally smoke in a business, but i don't know for sure |
| 15:25.51 | Katty | tompaw: i DO know, that you can absolutely REEK of smoke without smoking inside a build tho. |
| 15:26.02 | tompaw | yep, you're right. |
| 15:26.38 | Katty | i don't have an issue with someone smoking. |
| 15:26.46 | Katty | i do have an issue with the smells makes you want to throw up :< |
| 15:27.13 | tompaw | Trow up at them, they should get the message. |
| 15:27.47 | Katty | tompaw: that's not very polite :< |
| 15:28.12 | tompaw | Why? Cause it feels and smells bad. Just like the cig smell. |
| 15:28.15 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:28.34 | Katty | cause i am a polite person. |
| 15:28.45 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
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| 15:28.58 | Katty | and while i won't say anything to the person, and be as pleasant as possible...that doesn't mean i don't wish they'd stop making my office smell icky :P |
| 15:29.14 | tompaw | Katty: that's not very polite :< |
| 15:29.31 | Katty | well it's a lot more polite than telling him he stinks. |
| 15:29.32 | tompaw | Because of your passiveness, that person will keep polluting other people's private spaces. |
| 15:30.41 | tompaw | Maybe try writing him a mildly suggestive poem? Like: "Roses are pink, you f***g stink" |
| 15:30.56 | Katty | that's an awful idea. |
| 15:31.01 | Katty | you have no idea what kind of day they're dealing with. |
| 15:31.13 | Katty | their wife maybe in the hospital. |
| 15:31.17 | Katty | a child may have just been diagnosed with cancer. |
| 15:31.23 | Katty | they may have a huge bill they're not sure how they're going to pay |
| 15:31.37 | Katty | saying insensitive things just isn't using your noodle. |
| 15:31.38 | tompaw | You know what? you might be right, but I thought we were talking about a regular here. |
| 15:32.15 | tompaw | The way I see it, if you're passive and intravertic about it, you'll just pass the negative energy on your family&friends, as it needs to blow sooner or later. |
| 15:33.01 | Katty | i rant quite frequently to a very good listener. |
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| 15:34.13 | tompaw | And what do you think he/she does with with this energy? |
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| 15:34.38 | Katty | it gets his mind off his day for a little bit (= |
| 15:34.45 | Katty | and then he rants to me. |
| 15:35.24 | tompaw | Besides, you couldn't have picked up worst example than cancer. My best man is a surgeon and we used to talk a lot why THC is illegal in most places while the cigs are. |
| 15:35.48 | Katty | you know you're being awfully negative this morning. |
| 15:36.05 | tompaw | The guy does know a lot about medicine and science and there is only one answer: cigs are legal to provide a relief for a social security system. |
| 15:36.09 | Katty | are you having a bad day? |
| 15:36.26 | tompaw | Yep, IRS is having a long and painful control at my company. |
| 15:36.34 | Katty | i' msorry to hear that )= |
| 15:36.37 | Katty | i hope it gets better for you soon,. |
| 15:36.40 | tompaw | You're a great listener! |
| 15:36.45 | tompaw | Already feel better, thanks! :-) |
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| 15:38.26 | tompaw | Also, I'm trying to set up a PBX in Panama City and the latency between the agents office and the server house is twice the latency between that office and the US :/ |
| 15:38.32 | tompaw | Even though they're in the same town. |
| 15:39.15 | tompaw | BTW - does anyone if G729 patents are valid in Panama? |
| 15:40.30 | tompaw | I mean do I have to pay the royalties in Panama or is it free to use there? Sipro Lab website doesn't say a word about their wordlwide coverage. |
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| 15:50.44 | Qwell | tompaw: You'll need to consult an attorney specializing in IP laws in your country. |
| 15:51.03 | Qwell | Hint: They are expensive. |
| 15:51.12 | tompaw | :< |
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| 16:06.02 | voipsec123 | ERROR[12910] res_jabber.c: Malformed Jabber ID : phonesystem/asterisk (domain missing?) - JABBER: phonesystem INCOMING: <?xml version='1.0' encoding='UTF-8'?><stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client" from="nnnnn.xxxxxxxx.com" id="c24382c5" xml:lang="en" version="1.0"><stream:features><mechanisms xmlns="urn:ietf:params:xml:ns:xmpp-sasl"><mechanism>DIGEST-MD5</mechanism><mechanism>PLAIN</mechanism><mechanism>CRA |
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| 16:16.33 | *** join/#asterisk DigiDaz (~quassel@ks305112.kimsufi.com) |
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| 16:21.50 | DigiDaz | Hi all, I'm currently playing with asterisk in the cloud using version 1.8. I'm trying to use realtime connecting to either amazon or rackspace databases. They use really long hostnames ie > 50 characters. I seem unable to connect using the actual hostname but IP is fine. Does anyone know if there is a restriction on the length of hostname? |
| 16:22.17 | voipsec123 | DigiDaz: what protocol? |
| 16:22.29 | DigiDaz | Will be TCP I'm assuming |
| 16:22.37 | voipsec123 | TCP SIP? |
| 16:23.11 | DigiDaz | No, UDP for my telephony but it will be TCP connecting to the DB? |
| 16:23.22 | WIMPy | tompaw: Why don't you just ask them? |
| 16:23.31 | voipsec123 | ohh. your asking about realtime hostname to DB? |
| 16:23.33 | DigiDaz | Like I say, using the IP is fine |
| 16:23.44 | DigiDaz | Yes, sorry if I wasn't clear |
| 16:23.59 | voipsec123 | i shortcircuit like C |
| 16:24.07 | voipsec123 | no worries. let me see if I can dig up for a min |
| 16:24.07 | voipsec123 | sec |
| 16:24.16 | *** join/#asterisk jmls1 (~julian@host217-36-208-155.in-addr.btopenworld.com) |
| 16:24.27 | jmls1 | afternoon all |
| 16:24.33 | DigiDaz | Hi |
| 16:25.06 | jmls1 | does anyone know of a windows-based product that can do callerid lookup on facebook/googleapps/outlook etc ? |
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| 16:26.53 | voipsec123 | DigiDaz: how many chars fmi? |
| 16:29.01 | DigiDaz | I think the amazon one was 52 |
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| 16:29.42 | DigiDaz | Rackspace 3902da1ae83415df44252adddcb573067d6c782a.rackspaceclouddb.com :( |
| 16:30.51 | voipsec123 | wow. |
| 16:30.52 | tompaw | WIMPy: Sipro? |
| 16:31.08 | voipsec123 | lol. does nslookup work on local box for that fqdn? |
| 16:31.08 | DigiDaz | I'm using a2billing, it seems a very easy way to scale, multiple asterisk boxes to same db |
| 16:33.15 | paulc | 'c |
| 16:33.27 | WIMPy | tompaw: Yes |
| 16:33.43 | tompaw | Somehow it didn't occur to me ;) |
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| 16:34.27 | WIMPy | tompaw: I'm sure they won;t like it and will try to avoid answering, but you can always ask for proof. |
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| 17:05.29 | aboven | Which "dtmfmode" should I choose (rfc283, inband, info) for cellphone calls ? |
| 17:06.12 | [TK]D-Fender | Whatever matches what is being provided by the other end |
| 17:06.21 | [TK]D-Fender | Thre is no such setting for "cellphones". |
| 17:06.28 | [TK]D-Fender | There is only "what thte other SIP end is using |
| 17:07.00 | Katty | warms cold paws on [TK]D-Fender |
| 17:07.51 | gavimobile | I would like to create a dialplan rule that when I call, it will ask me what number I would like to dial, then it will dial my cell phone and connect me to the number which I entered. here is what I have so far however dialplan stops once the call is forwarded to my cellphone. the dialplan than continues after I hangup http://pastebin.com/hdkevekN |
| 17:08.39 | WIMPy | You need to place a new call. See the many forms of Originate. |
| 17:09.19 | gavimobile | WIMPy: was that for me |
| 17:09.27 | WIMPy | yes |
| 17:11.17 | gavimobile | WIMPy: I didn't follow |
| 17:12.19 | gavimobile | maybe you can give me an example? |
| 17:12.22 | WIMPy | You can't do it in one call. |
| 17:12.47 | gavimobile | WIMPy: that's fine, I know that I cannt |
| 17:12.56 | WIMPy | You need to end that call and then place another call. |
| 17:13.02 | gavimobile | but I have more than 1 channel. I can make 2 calls than combine them |
| 17:13.26 | gavimobile | WIMPy: I can do this manually with zoiper. why can't I configure asterisk to do this for me |
| 17:13.39 | WIMPy | "One call" as in the dialplan. |
| 17:13.46 | WIMPy | You can, but not in one call. |
| 17:13.55 | gavimobile | with zoiper I enter my cell number, once the call connectes I enter a tranfer number and it connects me |
| 17:14.20 | WIMPy | google for callback, maybe together with callthroough to get better results. |
| 17:14.46 | WIMPy | That's what you have to do in your dialplan as well. |
| 17:14.53 | gavimobile | WIMPy: im not following you man |
| 17:15.30 | youjelly | gavimobile: what are you trying to do |
| 17:15.37 | gavimobile | In general I won't be using google. most of my calls are international |
| 17:16.21 | gavimobile | but when I enter the number I wish to dial, I configured my dialplan to choose the best outbound route for the call depending on the number |
| 17:16.25 | WIMPy | "google" was used as a verb, mening to use a search engine. |
| 17:16.35 | gavimobile | youjelly: I am trying to build a dialplan application which will ask me to dial a number, then the dialplan will call my cell and connect me to the specified number |
| 17:16.50 | gavimobile | WIMPy: im not sure what to search for |
| 17:17.10 | WIMPy | Then read what I wrote. |
| 17:17.11 | gavimobile | I was hoping my dialplan was not to far off, I just might need to add or change something small |
| 17:17.40 | gavimobile | ahh so callback is the name of an application? along with callthrough? |
| 17:17.51 | gavimobile | these are applications which you are suggesting me to readup about? |
| 17:18.17 | WIMPy | I'd call it service, but yes. |
| 17:18.36 | gavimobile | WIMPy: I see, ok.. sorry for the misunderstanding |
| 17:18.50 | gavimobile | im using 1.8, which of the two would better suite me? |
| 17:18.59 | gavimobile | I can find that out on my own actually |
| 17:19.01 | gavimobile | its ok |
| 17:19.40 | WIMPy | You need callback. Callthough would be only one call. |
| 17:20.16 | gavimobile | is it called queue callback? |
| 17:20.21 | gavimobile | http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback |
| 17:20.39 | WIMPy | No queues. |
| 17:20.49 | WIMPy | That's something completely different. |
| 17:20.58 | youjelly | calll files, local channels |
| 17:21.03 | youjelly | hangup extension |
| 17:21.09 | youjelly | there's lots of ways to implement |
| 17:21.26 | WIMPy | indeed |
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| 17:22.29 | gavimobile | still looking |
| 17:22.39 | gavimobile | im trying to stay away from databases and call files |
| 17:24.10 | WIMPy | You can priginate via files, *CLI, AMI or dialplan. |
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| 18:13.33 | gavimobile | maybe I can use disa |
| 18:15.02 | kannan | hello, once a SIP channel is bridged is it possible to allow audio of one channel only to be whisper to another sip channnel? So if X and Y are on a call, Z will be muted and hear only X, and not Y |
| 18:16.44 | [TK]D-Fender | gavimobile, You still ahve to make * call your phone. that is Originate or Call-files |
| 18:17.52 | gavimobile | thanks [TK]D-Fender ill check up on thoese applications |
| 18:19.34 | [TK]D-Fender | gavimobile, Or, if you are triggering this from another call (which it looks like you're doing), there is a 3rd option : Dial() your cell with the option to split the 2 parties to different spots in the dialplan once answered. |
| 18:20.13 | kannan | i see chanSpy with 'o' option [o - Only listen to audio coming from this channel. from https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ExtenSpy]. is this fit in with the the above scenario? |
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| 18:21.59 | kannan | or will both channels audio be heard, since the call is bridged |
| 18:22.24 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-lkqpiswpjvmvzsmo) |
| 18:22.51 | [TK]D-Fender | You are spying one channel. |
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| 18:23.10 | [TK]D-Fender | that is what you're hooked to. so what it sends in not what it receives.... |
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| 18:25.01 | kannan | [TK]D-Fender, thanks. i will test. |
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| 18:28.53 | youjelly | kannan: short answer, no |
| 18:29.00 | youjelly | even with chanspy |
| 18:29.20 | kannan | youjelly, oh o. i thought it was a yes, |
| 18:29.47 | youjelly | when its bridged, you can't mute a particular person |
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| 18:30.26 | youjelly | I mean, you can, but then the person would be muted for all other channels |
| 18:30.35 | youjelly | not for a particular channel |
| 18:31.04 | youjelly | I've tried pretty much everything, then resorted to writing own module :) |
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| 18:35.13 | kannan | youjelly |
| 18:35.16 | kannan | thanks |
| 18:36.20 | kannan | youjelly, it seems to work for me. at first sight |
| 18:37.04 | kannan | i have chanspy on DAHDi with 'qo' ; it is not playing the other channel audio, but only the one i am spying on |
| 18:37.15 | kannan | i havent a way to test with sip tho |
| 18:37.18 | kannan | yet |
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| 18:48.09 | kannan | ok chanspy(DAHDI-xx,qo) works fine, i can hear the channel i spy on but no audio from other channel |
| 18:48.25 | kannan | i am sure it must be same for SIP, but i have to check |
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| 18:54.25 | ChannelZ | kannan: what were you expecting using 'o'? |
| 18:54.34 | McBoingBo | I have a user who is getting their calls dropped/destroyed when someone calls their line while they are on it, only seems to be the 1 set right now |
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| 18:55.25 | kannan | 'o' means audio of both channels is not heard, only the channel i spy on |
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| 18:56.27 | ChannelZ | Yeah.. and weren't you complaining that you 'couldn't hear the other end'? Maybe I missed the original query |
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| 18:57.03 | ChannelZ | I'm wandering in and out so probably so |
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| 18:58.10 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 19:03.28 | aboven | exten => _.,1,AGI(agi:async) -> is this only redirecting numbers ? |
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| 19:07.29 | kannan | ChannelZ, sorry was on call. the original query was how to get only one channel (in a call) to whisper to another, X on call with Y, Z should be muted and hear on X ,but not Y |
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| 19:08.10 | kannan | thats on SIP, i dont have sip on the 1.8+ server to test, but i tested on dahdi and assume SIP will work same way with ChenSpy |
| 19:08.48 | kannan | so thats basically giving the result i want to have really |
| 19:09.24 | ChannelZ | If the media stream is running through Asterisk, yes it shouldn't work any differently |
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| 19:10.13 | ChannelZ | hmm freenode is a little sick to its stomach today |
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| 19:13.59 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 19:14.07 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 19:20.54 | bn-7bc | hmm isa there any way to stop asterisk creating cdr on a per call basis? |
| 19:24.53 | McBoingBo | not sure whats happening here, can someone help. Phone destroys previous call when another call comes in. User A calls User B, User C calls User A, User A cannot hear User B anymore, User B can hear User A....took a looksy at sip debug but not seeing anything obvious |
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| 19:36.04 | ageis | hi all |
| 19:36.12 | ageis | we are running asterisk 1.6.2 on grandstream phones |
| 19:36.43 | ageis | have it set so multiple accounts are rung when a certain extension is dialed |
| 19:37.00 | ageis | I wondered if there was a way to get the phone display to show which extension was dialed |
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| 19:39.40 | [TK]D-Fender | ageis, Change the callerid before you dial. |
| 19:39.56 | carrar | Swap the extension with the caller id name field |
| 19:40.07 | carrar | wel not swap |
| 19:40.07 | carrar | but add too |
| 19:40.07 | carrar | or replace |
| 19:40.32 | carrar | set the callerid name to the extension |
| 19:40.46 | ageis | ahh |
| 19:41.18 | ageis | would this be done in one line in extensions.conf ? |
| 19:41.27 | ageis | can you give me an example of what it would look like? |
| 19:41.51 | carrar | Set(CALLERID(name)=${EXTEN}) |
| 19:42.17 | ageis | could I just append it? |
| 19:42.21 | ageis | so can keep callerid data as well? |
| 19:42.27 | [TK]D-Fender | ageis, use the name in the name |
| 19:42.33 | [TK]D-Fender | </yodawg> |
| 19:42.59 | ageis | so name is an extra field I can use |
| 19:43.02 | ageis | and it will still show the number |
| 19:43.03 | ageis | <PROTECTED> |
| 19:43.29 | carrar | exten => _XXXX,1,Set(CALLERID(name)=${EXTEN}) |
| 19:43.58 | carrar | replace _XXXX with whatever extension this is for |
| 19:44.07 | carrar | no |
| 19:44.45 | carrar | depends on the phone |
| 19:45.49 | ageis | has a rather large display |
| 19:46.01 | ageis | can I put CALLERID(name) and ${EXTEN} together ? |
| 19:46.54 | youjelly | kannan: I was trying something like this, http://pastebin.ca/2207315 |
| 19:47.01 | youjelly | didn't work for me |
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| 19:48.38 | kannan | youjelly, i will see |
| 19:48.39 | carrar | some phones may display both the caller ID name and number |
| 19:48.39 | carrar | in that case yes |
| 19:48.39 | carrar | the name will be the extension dialed |
| 19:48.39 | carrar | the number will be the original caller id number |
| 19:48.39 | carrar | try it out |
| 19:48.39 | carrar | have fun |
| 19:49.41 | ageis | carrar: but since we have our phones identified internally |
| 19:49.47 | ageis | carrar: it would help to preserve the names as well |
| 19:50.00 | McBoingBo | Phone destroys previous call when another call comes in. User A calls User B, User C calls User A, User A cannot hear User B anymore, User B can hear User A, say whaaaaat? |
| 19:50.30 | youjelly | basically I was trying to find a way to selectively mute a particular participant in the conference from all other participants |
| 19:51.36 | youjelly | actually, not all, like A,B and C are on the conf, A should hear C but B shouldn't, but C should be able to hear both A and B |
| 19:51.49 | kannan | youjelly, i see |
| 19:52.08 | carrar | ageis, how much screen realestate do you have? |
| 19:52.16 | youjelly | but ran into some issues due to channel locking probably |
| 19:52.22 | youjelly | internals |
| 19:53.11 | carrar | exten => 1234,1,Set(CALLERID(name)=${CALLERID(name)} ${EXTEN}) |
| 19:53.38 | ageis | carrar like 2x5" |
| 19:53.44 | ageis | yeah I was wondering how to append variables to eachother |
| 19:53.49 | ageis | carrar thanks a lot |
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| 20:00.05 | nickgaw | Hi, Do Asus netbooks have any bios settings I need to change to stop the system going to sleep when running pbx in a flash or any other linux distribution? |
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| 20:02.02 | carrar | ${ASUS(bios)=nosleep} |
| 20:02.09 | youjelly | lol |
| 20:02.37 | nickgaw | how do I change this? |
| 20:02.50 | carrar | Call your PC Admin |
| 20:03.01 | eduzimrs | Hi, is it possible to hear clearly a call using MixMonitor with SRTP media stream? |
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| 20:04.33 | nickgaw | is it a bios setting I need to change as i first was thinking it was a windows issue but after installing linux after of course backing everything up I don't think that is the issue? |
| 20:04.57 | [TK]D-Fender | nickgaw, Wrong channel... go ask in a generic distro support channel.... |
| 20:05.13 | carrar | 1) Check the bios yourself |
| 20:05.25 | nickgaw | I will look more |
| 20:05.26 | carrar | 2) make sure your OS doesn't have shutdown enabled |
| 20:05.36 | carrar | aka shutdown for when you are on battery power |
| 20:05.37 | carrar | or whatenow |
| 20:05.40 | carrar | inactivity |
| 20:06.02 | nickgaw | the system is always plugd in as I am trying to convert it into a sip asterisk server for my house. |
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| 20:06.31 | carrar | maybe you are loosing power |
| 20:06.45 | youjelly | unplug it and plug it into the power lines |
| 20:06.48 | carrar | possibly a bad pwoer supply |
| 20:06.51 | youjelly | high voltage |
| 20:07.02 | nickgaw | for a hardware wireless sip phone that is like a normal land line but does sip over a wireless connection what phone does anyone here suggest? |
| 20:07.24 | McBoingBo | anyway, this is my sip debug log for the problem conversation where User A calls User B and User B cannot hear User A anymore as soon as another call comes into User B, call is not dropped, just no sound. http://pastebin.com/feerfCkd |
| 20:07.30 | carrar | polycom spectralink |
| 20:07.49 | carrar | or for better quality get a DECT SIP handset |
| 20:08.12 | nickgaw | I am looking for the sip handsets but do any of them have speaker phones? |
| 20:08.19 | carrar | yes |
| 20:08.51 | nickgaw | what good sip handsets do you suggest? |
| 20:09.48 | carrar | Aastra MBU-400 |
| 20:11.30 | nickgaw | can asterisk run if I have an external dynamic ip address? |
| 20:11.44 | carrar | yes |
| 20:11.58 | nickgaw | with google voice? |
| 20:12.32 | carrar | I don't use google voice, someone else can answer you |
| 20:13.07 | nickgaw | but can companies that provide the DID's and allow you to make normal phone calls work with a dynamic ip address? |
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| 20:22.05 | kresp0 | you dont need fixed IP for DID to work, nickgaw |
| 20:22.40 | kresp0 | But if you have problems, you might try to use some free dynamic dns services |
| 20:24.25 | nickgaw | so all providors are pritty much supporting dynamic ip addresses? |
| 20:25.41 | keith4 | registering with them sort-of gets around that problem |
| 20:25.47 | keith4 | just like using dynamics with phones |
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| 20:31.38 | Qwell | glares at FreeNode |
| 20:34.40 | citywok | this is the second time asterisk has crashed this same way, and nothing has changed: ERROR[7919] astobj2.c: refcount -1 on object 0x8a7a2f8 |
| 20:34.55 | citywok | does anybody have any ideas? i'm recompiling with debugging and will catch a core dump the next time it happens. |
| 20:35.37 | citywok | it goes all the way to refcount -728 on this one |
| 20:44.39 | ageis | carrar: I want to set the callerid before i dial, correct? |
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| 20:47.25 | Dovid | had anyone here gotten mplayer working with Asterisk? |
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| 20:57.16 | RackSmurf | It never stops.. |
| 20:57.42 | RackSmurf | *now + dpma + d50 won't play nice. *sigh* |
| 20:58.27 | RackSmurf | ls |
| 20:58.44 | Qwell | . .. g0ats regular |
| 20:59.17 | RackSmurf | Qwell: Any idea on potential causes for a d50 firmware update failing? |
| 20:59.39 | Qwell | not off the top of my head |
| 21:02.17 | Qwell | call support, or bug me about it tomorrow |
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