| 00:01.26 | TechSmurf | netbook? |
| 00:03.08 | pyther | hehe although that would work, not what I'm looking for |
| 00:03.47 | TechSmurf | http://www.amazon.com/Gateway-LT2016u-Netbook-Windows-Compatible/dp/B003TPKDY6/ |
| 00:03.51 | TechSmurf | for sake of mention |
| 00:04.05 | TechSmurf | goes back to trying to find something cheaper without monitor/kb built in... |
| 00:04.18 | pyther | I'd love one of these: http://www.ebay.com/itm/New-OEM-Supermicro-X7SPA-H-O-Atom-D510-Intel-ICH9R-RAID-V-2GbE-Mini-ITX-/320951231004?pt=Motherboards&hash=item4aba2f221c |
| 00:04.31 | pyther | way to expensive for an asterisks box |
| 00:04.41 | TechSmurf | yeah... |
| 00:07.41 | pyther | hell this is almost perfect: http://www.newegg.com/Product/Product.aspx?Item=N82E16856119067 |
| 00:09.05 | pyther | no serial though :( |
| 00:09.58 | Dar1us | USB serial adapter? :) |
| 00:10.48 | pyther | don't know how reliable they are, probably wouldn't work with grub |
| 00:11.15 | Dar1us | oh for a serial console? |
| 00:11.26 | pyther | yah, I don't have a monitor for it |
| 00:11.41 | pyther | granted with hdmi it would be pretty easy to physically move when there is an issue |
| 00:11.45 | Dar1us | a USB serial adapter won't work for that |
| 00:12.05 | Dar1us | you could get one with IMPI, would bump the price up significantly though :( |
| 00:12.33 | pyther | yes, the super micros are super nice, but almost $230 for the board |
| 00:12.55 | Dar1us | yeah |
| 00:13.16 | Dar1us | for the price differential you could almost buy a cheap LCD monitor :) |
| 00:13.53 | pyther | Part of me really wants to buy this case: http://www.newegg.com/Product/Product.aspx?Item=N82E16811345016 |
| 00:15.03 | Dar1us | looks pretty neat |
| 00:15.18 | pyther | Trying like hell to justify the cost of a system |
| 00:15.30 | TechSmurf | heh |
| 00:15.31 | pyther | one could argue I could use it for storage and asteriks |
| 00:15.38 | Dar1us | I was going to say |
| 00:15.45 | Dar1us | you could make a NAS box which also runs asterisk |
| 00:16.06 | pyther | dedicate disk for asterisks would work, and with cgroups I should be able to setup the system to give asterisk 100% priority |
| 00:16.15 | pyther | not that it requires a hell of a lot |
| 00:16.39 | Dar1us | why would you need dedicated disks? |
| 00:16.50 | pyther | reduce I/O |
| 00:17.02 | Dar1us | I doubt it's necessary unless you have assloads of users |
| 00:17.05 | pyther | I'd probably get a 32GB SSD with the OS on it, and then traditional disks for data |
| 00:18.02 | pyther | that build would probably be $250 :( |
| 00:20.18 | Curs0r | Well the client flatly refused to switch to flowroute. Looks like I'll be continuing to wonder why the dahdi card stops working seemingly at random |
| 00:20.36 | pyther | Curs0r: :( |
| 00:20.43 | pyther | I hope they are reliable, getting my number ported to them |
| 00:22.15 | Curs0r | Strangest thing, sometime's the card just doesn't acknowledge incoming and it routes to their analog provider's voicemail. I restart and it's fine |
| 00:37.20 | pyther | hell that is a tempting deal, that system with free 4GB ram |
| 00:39.19 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
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| 00:53.00 | bananapie | Anyone know what unit of measure is used for the rtpc data dumped after a SIP call ? I looked and can't find anything. I am wondering specifically about "avgrxjitter", Is this in seconds? |
| 00:57.24 | Dar1us | what's a typical value |
| 00:57.25 | Dar1us | ? |
| 01:01.04 | bananapie | 0.000420 is a sample value on voip-info |
| 01:02.35 | Dar1us | http://www.packetizer.com/rfc/rfc3611/ says its in the same units as the timestamp |
| 01:03.21 | bananapie | Wonderful, thanks :) |
| 01:04.17 | Dar1us | ... as the RTP timestamp I should say |
| 01:04.17 | Dar1us | OK |
| 01:14.20 | bananapie | are these stats collected before or after the generic jitter buffer is applied ? |
| 01:14.44 | bananapie | because the values I am seeing don't make sense. |
| 01:15.14 | Dar1us | dunno sorry, I only looked through the code and the RFC a little |
| 01:15.51 | bananapie | I looked through the code, but I barely understand C |
| 01:25.01 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-7-26.se.biz.rr.com) |
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| 01:40.55 | drmessano | What package is needed so when I run make menuselect I get the color, tabled menu vs the non-color list menu? |
| 01:43.01 | WIMPy | newt |
| 01:43.28 | WIMPy | So probably newt-dev or something. |
| 01:46.14 | drmessano | w00t... That did it.. libnewt-dev |
| 01:46.16 | drmessano | Thank you |
| 01:47.12 | drmessano | At some point I had that and installed on one box, but didn't document it.. Annoyed me since. |
| 01:47.53 | WIMPy | I don't like how the newt UI works. |
| 01:48.37 | drmessano | What dont you like? |
| 01:49.29 | WIMPy | I always press the wrong keys. doesn't happen with the ncurses version. |
| 01:50.05 | drmessano | space vs enter and all |
| 01:50.16 | WIMPy | Yes |
| 01:51.58 | drmessano | I like that the enter key selects and jumps to the next item. Handy for the MOH and Sound packages |
| 01:52.58 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
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| 02:00.29 | *** join/#asterisk boch (~boch@186.182.121.168) |
| 02:00.35 | boch | good night all |
| 02:01.41 | drmessano | You joined to say goodnight? |
| 02:01.43 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 02:04.44 | *** part/#asterisk newtonr (~newtonr@173-25-121-44.client.mchsi.com) |
| 02:06.35 | WIMPy | It might have been a greeting. |
| 02:10.44 | pyther | Anyone have any experience with preloading all the necessary asterik files into ram on boot? |
| 02:11.43 | Curs0r | How odd, I have just had another occurrence in which the pbx server was not responding to an incoming pstn call until I restarted it. Can't figure out why it keeps getting stuck like that |
| 02:12.10 | WIMPy | What kind of hardware? |
| 02:12.35 | Curs0r | TDM410P w/o the hwec |
| 02:13.24 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-18-80.bois.qwest.net) |
| 02:16.35 | boch | drmessano, i am sorry for wishing you a good night, please ignore me |
| 02:17.49 | boch | for the rest, have a good night |
| 02:29.25 | *** join/#asterisk sustav (~vpp@76.73.166.16) |
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| 02:42.06 | Curs0r | Interesting. I called it 4 times and watched the console. It used each dahdi port, with varying success passing along to voice mail after the appropriate amount of time. Now, it's refusing to accept incoming calls again. Perhaps it's not releasing the dahdi channel when I hang up? |
| 02:46.04 | WIMPy | "varying success"? |
| 02:46.49 | Curs0r | Channel 3 picked up and should have sent to voicemail but instead it rang over to channel 2 to deliver voicemail. I found that odd |
| 02:47.18 | Curs0r | Channel 1 picked up and should have delivered voicemail as the others do but was dead air |
| 02:48.33 | Curs0r | And that exhausted my supply of channels and their number hunt started sending the calls into their provider's voicemail system until I rebooted |
| 02:48.55 | WIMPy | Analog is evil! |
| 02:49.02 | Curs0r | Don't I know it |
| 02:49.34 | Curs0r | I'm having a sit down with them tomorrow to try and talk them into sip trunking again |
| 02:51.07 | Curs0r | Turns out these analog jacks are from a freaking cable provider so I'm like uhhh... can we cut some deadwood out here? |
| 02:51.29 | WIMPy | So it's probably SIP already? |
| 02:51.36 | Curs0r | lol probably |
| 02:52.04 | WIMPy | Ask Dr Google how to get the accounts. |
| 02:53.11 | Curs0r | So this retarded giant grey box that's feeding 4 jacks that I found behind their copier, I can cut that out of the loop without even changing their service plan? |
| 02:53.43 | WIMPy | Maybe. |
| 02:53.53 | WIMPy | Quite possible. |
| 02:53.59 | Curs0r | That would be something... |
| 03:12.44 | carrar | crap, obama is in town messing up our traffic |
| 03:13.31 | *** join/#asterisk sustav (~vpp@76.73.166.16) |
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| 03:40.50 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 05:01.15 | *** join/#asterisk awk (~awk@trancehouse.co.za) |
| 05:01.39 | awk | Hi, what is the best way in asterisk to set a keep alive on an extension, it keeps dropped off, its coming in from a external source. |
| 05:01.49 | awk | Would this be done on the client or is there a asterisk (extension) option |
| 05:02.05 | awk | I was trying to use the clients to send option packets as well as udp and it still drops off |
| 05:02.14 | WIMPy | You can do both. |
| 05:02.19 | awk | once I make a call it re-connects, but that doesn't help someone calling in |
| 05:02.33 | WIMPy | qualifyfreq |
| 05:02.55 | WIMPy | But the client may have a real keepalive option. |
| 05:03.09 | awk | thanks, let me look at that variable. |
| 05:03.31 | awk | that isn't a global sip.conf option its a [exten] option |
| 05:03.33 | awk | ? |
| 05:03.47 | WIMPy | both I think |
| 05:04.04 | awk | I have qualify time in ms set to 2000 |
| 05:04.09 | awk | What should I try change that too? |
| 05:04.31 | WIMPy | That's the timeout. |
| 05:04.57 | WIMPy | qualifyfreq sets how often to send OPTIONs. |
| 05:05.17 | awk | ahh |
| 05:05.26 | awk | smart boi |
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| 06:30.44 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:30.46 | schmidts | good morning |
| 06:32.19 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 06:32.22 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 06:32.25 | v0lZy | lo |
| 06:38.10 | *** join/#asterisk _omer (~omer@116.71.191.183) |
| 06:40.15 | _omer | my asterisk config files are not located in default folder. I have changed asterisk config folder in /etc/amportal.conf ... when I do "amportal restart" I get error ... "chown /etc/asterisk/ ........." something like that .... I just want to know why amportal is still looking in default asterisk folder? |
| 06:42.36 | *** join/#asterisk doolittlework (~doolittle@41-134-22-14.dsl.mweb.co.za) |
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| 06:50.22 | v0lZy | guys, i have some issues regarding caller id |
| 06:50.39 | v0lZy | is there a way i can code something in asterisk that will show the id when i press a key sequence? |
| 06:50.44 | v0lZy | like ### or something? |
| 06:55.29 | doolittlework | v0lZy: what do you want to do? |
| 06:55.31 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:57.35 | v0lZy | doolittlework: my users want to see the caller-ID of the original caller after an attended transfer |
| 06:58.25 | v0lZy | this seems impossible because when doing an attended transfer, im actually calling as i would normally and only after the call is put through, i bridge whoever called me with whoever i transfered to |
| 06:59.46 | v0lZy | now as far as i understand, this has to do with sip headers etc |
| 06:59.59 | v0lZy | and when asterisk sends something to the phone, the phone shows it |
| 07:00.12 | doolittlework | v0lZy: just the set the ${CALLERID(all)} to the caller id received initialy |
| 07:02.33 | v0lZy | hm... how... and when? |
| 07:03.30 | v0lZy | right now im using |
| 07:03.37 | v0lZy | sendrpid = pai |
| 07:03.58 | v0lZy | sendrpid=pai i mean |
| 07:04.05 | v0lZy | and this works to a certain extent |
| 07:04.42 | v0lZy | the problem is when i do a call pickup |
| 07:04.45 | v0lZy | or an attended transfer |
| 07:04.51 | v0lZy | i dont see the original callers number |
| 07:06.05 | v0lZy | mybe also rpid_updated=yes? |
| 07:09.30 | v0lZy | rpid_update=yes i mean.. |
| 07:09.40 | doolittlework | v0lZy I think when you set the callerid(all) when the call starts it remains the same, i might be wrong but that is how i used it on one of my system |
| 07:10.17 | v0lZy | where do u set that? |
| 07:11.46 | doolittlework | in your dialplan when the call comes in from the carrier and internal dialplan |
| 07:13.59 | doolittlework | paste your incomming context then i can try and help |
| 07:14.08 | v0lZy | Ok |
| 07:14.22 | v0lZy | Hopefully I get this right. Im working my way backwards from a preconfigured dialplan |
| 07:14.31 | v0lZy | (askoziapbx) |
| 07:14.40 | v0lZy | it shoudl have [incoming] context? |
| 07:15.38 | v0lZy | i have a provider-incoming thing |
| 07:15.41 | v0lZy | i guess thats it? |
| 07:16.49 | doolittlework | what version of asterisk are you running |
| 07:17.32 | v0lZy | 1.8 |
| 07:17.33 | v0lZy | http://pastebin.com/iaWcKSrG |
| 07:18.37 | v0lZy | its a bit cryptic |
| 07:20.55 | v0lZy | basically this is my problem |
| 07:20.57 | v0lZy | http://www.freepbx.org/forum/freepbx/users/caller-id-is-lost-after-an-attended-transfer-or-for-directed-call-pickup |
| 07:20.58 | v0lZy | only on asterisk 1.8 |
| 07:21.21 | v0lZy | Asterisk 1.8.4.4 built by root @ on a i486 running Linux on 2012-04-25 16:59:15 UTC |
| 07:23.20 | doolittlework | are you using freepbx? |
| 07:24.14 | doolittlework | The link to freepbx refers to astra phones |
| 07:25.36 | v0lZy | hm |
| 07:25.46 | v0lZy | no,a sterisk but |
| 07:25.49 | v0lZy | asterisk* |
| 07:25.54 | v0lZy | but the description of the problem |
| 07:26.08 | v0lZy | matches my predicament |
| 07:26.17 | v0lZy | DCC doesnt work :| |
| 07:27.23 | doolittlework | i see |
| 07:27.52 | doolittlework | i am lloking through my notes, i know i had this before, what phones are you using? |
| 07:28.39 | v0lZy | one aastra phone |
| 07:28.44 | v0lZy | and many draytek phones |
| 07:28.48 | v0lZy | draytek vigorphone |
| 07:28.56 | v0lZy | But reading on this on forums for earlier asterisk version |
| 07:29.04 | v0lZy | they say the problem is sending a reinvite |
| 07:29.44 | *** join/#asterisk Vince-0 (c4d79abe@gateway/web/freenode/ip.196.215.154.190) |
| 07:30.16 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:b5ff:ff9d:f7f3:d11f) |
| 07:30.22 | v0lZy | http://wiki.snom.com/FAQ/Why_is_the_callerid_not_changed_after_transfer_with_asterisk%3F |
| 07:31.25 | v0lZy | lots of people report this |
| 07:31.33 | v0lZy | i have sendrpid=pai arleady |
| 07:32.23 | ChannelZ | I don't think this is a "problem" - in an attended transfer, the person transferring is making a new call to whomever they're transferring to. |
| 07:33.56 | ChannelZ | (so the Caller ID will be that of the person transferring, not the call being transferred.) |
| 07:36.41 | doolittlework | my take on this is asterisk is not a sip proxy so it does not forward traffic(callerid) Each call is kept separate from the other, so when you do an attended transfer and finish it - the other side doesn't know it occured |
| 07:37.59 | doolittlework | I had it working before where i used the cdr(src) to pass onto the next call |
| 07:40.24 | *** join/#asterisk youjelly (~bwahahaha@182.177.58.104) |
| 07:42.06 | *** join/#asterisk AlHafoudh (~alhafoudh@85.248.11.120) |
| 07:42.07 | AlHafoudh | hi all |
| 07:42.13 | doolittlework | hi |
| 07:43.05 | AlHafoudh | my sip provider wants me to send From: header like "From: "Anonymous"<sip:anonymous@anonymous.invalid>" to enable CLIR for the call. Is it possible with asterisk? I tried SipRemoveHeader and SipAddHeader but it does not remove the From: and adds the second one |
| 07:43.23 | *** join/#asterisk linocisco (c186f20c@gateway/web/freenode/ip.193.134.242.12) |
| 07:44.23 | doolittlework | looks like its all callerid problems today |
| 07:44.45 | ChannelZ | heh yeah |
| 07:44.54 | linocisco | hi all i have android phone with CSIPsimple installed. I dont know how to configure it with asterisk |
| 07:44.56 | doolittlework | v0lZy: try the f: forces callerid to be set based on a dialplan "hint" for the current channel. For example, some PSTNs don't allow callerids from other extensions than the ones that are assigned to you. in the Dial command |
| 07:45.52 | AlHafoudh | callerid! i havent tried that |
| 07:45.54 | ChannelZ | AlHafoudh: set the callerid info for the peer you use for them in sip.conf -- callerid=Foo <12345> |
| 07:46.48 | AlHafoudh | but that collides with P-Asserted-Identity i need to send :( |
| 07:47.20 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:47.32 | doolittlework | AlHafoudh : asterisk command SIPAddHeader(Header: Content) |
| 07:47.53 | v0lZy | ChannelZ: i agree its not a 'problem'... but my users think otherwise and im their bitch. |
| 07:48.10 | AlHafoudh | doolittlework: it wont allow mne to change From: like that |
| 07:48.11 | *** join/#asterisk asterisk_gurl (~asterisk_@116.214.96.99) |
| 07:48.48 | AlHafoudh | ChannelZ: i tried that, but it will not set the domain after "@" sign |
| 07:48.49 | linocisco | hi all i have android phone with CSIPsimple installed. I dont know how to configure it with asterisk |
| 07:49.43 | v0lZy | doolittlework: i dont understand (im not that knowledgable to asterisk).... f is a flag to use with dial command? |
| 07:50.00 | doolittlework | linocisco: is CSIPsimple asterisk on a android phone |
| 07:50.06 | doolittlework | v0lZy: yes |
| 07:50.56 | ChannelZ | AlHafoudh: I think asterisk will normally use your IP, they actually want it to say "anonymous.invalid" ???? |
| 07:51.15 | AlHafoudh | ChannelZ: yeah :( |
| 07:51.21 | ChannelZ | that's stupid as hell |
| 07:51.31 | ChannelZ | try setting fromdomain=anonymous.invalid |
| 07:51.31 | AlHafoudh | ChannelZ: its some freakin Huawei SBC |
| 07:51.49 | AlHafoudh | ChannelZ: can I set it in extensions? |
| 07:52.05 | v0lZy | doolittlework: so i have to modify my dialplan? |
| 07:52.08 | v0lZy | What part of it? |
| 07:52.10 | ChannelZ | don't think so. |
| 07:52.22 | v0lZy | its a huge dialplan |
| 07:52.43 | doolittlework | the part where you dial the extensions paste your extensions.conf file and i will have a look |
| 07:52.49 | ChannelZ | and actually the way fromdomain is described it might only work for non-configured peers.. hmm |
| 07:52.59 | asterisk_gurl | Hi guys, i am trying to make redfone working on my asterisk server, i have installed dahdi-linux, dahdi-tools,fonulator,libfb. I also configured redfone.conf, module and system.conf... But when I run dahdi_tool it is blank.. |
| 07:53.16 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:53.38 | ChannelZ | is that a TDM-over-Ethernet thing? |
| 07:53.39 | doolittlework | asterisk_gurl : what does dahdi-scan say |
| 07:53.45 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:53.51 | doolittlework | sorry dahdi_scan |
| 07:55.25 | doolittlework | asterisk_gurl : is that not a gateway, so i think it would not use dahdi |
| 07:55.31 | linocisco | doolittlework: sure |
| 07:55.33 | asterisk_gurl | When i also run dahdi_genconf and dahdi_hardware, it gives an error argument "dynamic-ethmf" isn't numeric in array elemnt at /urs/lib/perl/site_perl5/5.8.8/Dahdi.pm and Span.pm |
| 07:56.52 | ChannelZ | What did you put in your /etc/dahdi/system.conf ? |
| 07:57.57 | asterisk_gurl | Dynamic=ethmf,eth/MAC of REDFONE/0,24,0 |
| 07:58.36 | asterisk_gurl | Then when I run dahdi_genconf, it was written in different way but the mac of redfone is still correct |
| 07:59.20 | ChannelZ | not even sure genconf understands dynamics |
| 07:59.47 | ChannelZ | but shouldn't your 'eth' be 'eth0' or 'eth1' or whatever ethernet device it's connected to? Not just 'eth' |
| 07:59.57 | asterisk_gurl | Dahdi_scan span 1 = dummy; span 2 active yes alarms=red |
| 08:00.12 | AlHafoudh | ChannelZ: just fun note, it was sufficient to set "Privacy: id" |
| 08:00.26 | asterisk_gurl | Yeah, eth1 sorry, typo error here |
| 08:01.58 | ChannelZ | AlHafoudh: what version of DAHDI |
| 08:02.25 | ChannelZ | pp[d |
| 08:02.27 | ChannelZ | oops |
| 08:02.32 | ChannelZ | sorry that was directed to asterisk_gurl |
| 08:02.39 | asterisk_gurl | Hmm... What's the command to check version? Sorry im just new in astrisk |
| 08:02.41 | AlHafoudh | ;) |
| 08:02.55 | ChannelZ | asterisk_gurl: did you compile it yourself or is this a package? |
| 08:02.55 | *** join/#asterisk youjelly (~bwahahaha@182.177.49.98) |
| 08:03.20 | asterisk_gurl | I compile it my self |
| 08:03.38 | ChannelZ | so the source directory/tarball should say :) |
| 08:04.33 | ChannelZ | else "dahdi show version" on the asterisk console aught say |
| 08:04.43 | asterisk_gurl | I use dahdi-linux-redfone-2.1.0.4 and dahdi-tools-2.0.0 |
| 08:05.09 | ChannelZ | apparently that's a bug |
| 08:05.38 | asterisk_gurl | Ohhh... So what's the recommended version? |
| 08:05.45 | ChannelZ | anything over that :) |
| 08:05.49 | ChannelZ | http://bugs.elastix.org/print_bug_page.php?bug_id=87 |
| 08:05.57 | ChannelZ | for example |
| 08:07.46 | doolittlework | ChannelZ ahs changed his name to GOOGLE |
| 08:07.50 | ChannelZ | I don't actually know anything about these redfone devices, do they really require their own special DAHDI build? |
| 08:08.28 | asterisk_gurl | im not really sure because im new here in asterisk.. |
| 08:08.36 | ChannelZ | doolittlework: if only I got paid to do it |
| 08:08.58 | ChannelZ | well where did you get it in the first place? Is it spelled out in the redfone instructions or something? |
| 08:09.26 | ChannelZ | in any case you can configure the spans yourself regardless |
| 08:09.51 | asterisk_gurl | My collegue gave me this redfone device and asked me to configure it on the asterisk server.. |
| 08:10.03 | ChannelZ | nice. |
| 08:11.04 | ChannelZ | I dunno if it works or not but my DAHDI 2.6.x build anyway has the ethmf driver in it, I'm imagining you can maybe use off-the-shelf DAHDI for this. |
| 08:11.08 | asterisk_gurl | I tried to research some documentations how to install it and it says there need to download and compile those dahdi-linux, tools,fonulator and lib |
| 08:11.13 | ChannelZ | Just need to make sure you recompile Asterisk too though |
| 08:11.56 | ChannelZ | but I should imagine redfone would have modern documentation if they really have special requirements |
| 08:12.22 | ChannelZ | Alas I'm of not much help besides generalities since I"ve never touched one of the things. And it's past my bedtime |
| 08:13.09 | asterisk_gurl | How do i recompile asterisk? |
| 08:13.27 | asterisk_gurl | Does it mean i need to download something? |
| 08:14.35 | doolittlework | asterisk_gurl depends on how you installed it |
| 08:17.08 | asterisk_gurl | Ok,thanks |
| 08:21.53 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:27.55 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 08:53.53 | EmleyMoor | Is there a limit to the number of additional ring cadences you can have on DAHDI? Using Asterisk 1.6.2.9-2+squeeze6, not seeing a 7th cadence I tried to set |
| 08:56.35 | *** join/#asterisk _zoom_ (~Eissa@196.1.219.122) |
| 08:56.51 | EmleyMoor | Found an error in it |
| 08:56.58 | _zoom_ | hi, how to intergrate asterisk with sugarcrm ce |
| 08:57.00 | _zoom_ | ? |
| 08:57.23 | Faustov | properly! |
| 08:57.41 | EmleyMoor | Sorted. |
| 09:00.40 | *** join/#asterisk asterisk_gurl (~asterisk_@119.234.0.34) |
| 09:01.26 | asterisk_gurl | H, i just updated my dahdi to version 2.6.0 and now it gives me a kernel panic - not syncing: fetal exception in interrupt |
| 09:01.54 | asterisk_gurl | I am rebooting the server when i got this message |
| 09:02.57 | linocisco | hi all i have android phone with CSIPsimple installed. I dont know how to configure it with asterisk |
| 09:05.18 | _zoom_ | linocisco: whatever client your are using, at least you need to put 3 values domain, username, password |
| 09:06.10 | linocisco | ok |
| 09:06.27 | linocisco | _zoom_: how to configure in asterisk? |
| 09:07.39 | _zoom_ | linocisco: sip.conf |
| 09:07.46 | _zoom_ | you will find examples there |
| 09:08.20 | linocisco | _zoom_: any web link? I now have no access to sip.conf as i have not installed asterisknow yet. |
| 09:08.43 | linocisco | _zoom_: step by step screenshot based guide is more appreciated |
| 09:10.48 | _zoom_ | i really dont remember the exact steps on asterisknow or trix box, but at the end they edits /etc/asterisk/sip.conf |
| 09:14.48 | *** join/#asterisk youjelly (~bwahahaha@182.177.78.165) |
| 09:19.59 | EmleyMoor | Is there a field available for a SIP channel that holds the incoming IP address? |
| 09:21.40 | tuxx- | hey guys, anyone know what the interval is that asterisk uses the 'sanitysql' option in res_odbc.conf ? |
| 09:23.12 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 09:32.15 | rolandow | tuxx-: it doesn't reconnect? |
| 09:32.31 | rolandow | tuxx-: what pre-connect setting do you have? |
| 09:32.54 | tuxx- | pre-connect => yes |
| 09:33.18 | *** join/#asterisk danfromuk (~IceChat77@2.27.12.118) |
| 09:33.19 | rolandow | ok .. that should be fine from what i have read :) |
| 09:33.26 | tuxx- | well, i tried this morning |
| 09:33.31 | tuxx- | and the db connection was once again lost |
| 09:33.51 | tuxx- | so the first call failed again, hehe :) |
| 09:35.04 | rolandow | so does it automatically reconnect after the first failed call? |
| 09:35.41 | rolandow | because i'm thinking if it does, maybe you could generate a call file every hour to keep your connection alive |
| 09:36.11 | rolandow | it's a bit quick and dirty solution though |
| 09:39.26 | tuxx- | right |
| 09:39.30 | tuxx- | should work idd |
| 09:39.42 | tuxx- | and idd, its only the first call, after that it reconnects without a problem |
| 09:40.48 | tuxx- | i'll make a cronjob that makes a fakecall to work around this problem |
| 09:52.37 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.173) |
| 09:52.42 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:04.50 | linocisco | http://www.fivn.com/products/asterisk.html looks good |
| 10:12.11 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:21.25 | linocisco | where is the asterisk image for my WRT110 Linksys router?? I dont see any of them on openwrt.org |
| 10:23.04 | v0lZy | guys |
| 10:23.12 | v0lZy | any reasons why i cant pull out a global variable? |
| 10:23.14 | v0lZy | i set it |
| 10:23.25 | v0lZy | but then its empty later on, as if it wasnt set? |
| 10:24.34 | EmleyMoor | Can you pastebin the relevant bits of your dialplan? (paste.debian.net is a cood pastebin, there are others) |
| 10:26.16 | v0lZy | EmleyMoor: im working with call transfers and im trying to artifically do |
| 10:26.18 | v0lZy | exten => 44,n,Set(CALLERID(all)=${OUTSIDECALLERID}) |
| 10:26.27 | v0lZy | this is in my [incoming] context |
| 10:26.43 | v0lZy | in my provider context for incoming calls, i have |
| 10:26.57 | v0lZy | exten => 015809316,n,Set(GLOBAL(OUTSIDECALLERID=0${CALLERID(number)}) |
| 10:27.48 | EmleyMoor | v0lZy: It's not CALLERID(number), it's CALLERID(num) |
| 10:28.05 | v0lZy | exten => 015809316,n,Set(GLOBAL(OUTSIDECALLERID=0${CALLERID(number)}) |
| 10:28.06 | v0lZy | exten => 015809316,n,NoOP(********${OUTSIDECALLERID}********) |
| 10:28.09 | v0lZy | and in my cli |
| 10:28.10 | v0lZy | i get |
| 10:28.26 | v0lZy | <PROTECTED> |
| 10:28.28 | v0lZy | <PROTECTED> |
| 10:28.29 | v0lZy | <PROTECTED> |
| 10:28.35 | v0lZy | CALLERID(number) works ok |
| 10:28.45 | EmleyMoor | Hmmm... |
| 10:28.49 | v0lZy | but as u can see |
| 10:28.54 | v0lZy | im not picking up the variable |
| 10:29.01 | v0lZy | am i calling it wrong? |
| 10:29.14 | EmleyMoor | Hold on... |
| 10:29.23 | v0lZy | i have a sense that im fetching it wrong |
| 10:29.41 | v0lZy | ${OUTSIDECALLERID} i thought would pick it up, but it doesnt |
| 10:30.17 | EmleyMoor | Do you have an initial definition of OUTSIDECALLERID? |
| 10:31.18 | v0lZy | exten => 015809316,n,Set(GLOBAL(OUTSIDECALLERID=0${CALLERID(number)}) |
| 10:31.22 | v0lZy | this is the first time i mention it anywhere |
| 10:31.27 | EmleyMoor | That's not initial... |
| 10:31.57 | EmleyMoor | I'm also wondering if the imbalanced brackets in that line are affecting it |
| 10:31.57 | v0lZy | how do i do an initalization then? |
| 10:32.21 | v0lZy | ah |
| 10:32.23 | v0lZy | more brackets.. |
| 10:32.40 | EmleyMoor | See if an extra ) on the end helps. If not, put it in the [globals] section of your dialplan |
| 10:33.29 | v0lZy | the extra didnt do anythin gi think |
| 10:33.47 | v0lZy | what do i put there? |
| 10:33.58 | EmleyMoor | OUTSIDECALLERID=whatever |
| 10:35.30 | v0lZy | uh |
| 10:35.31 | v0lZy | ok |
| 10:35.36 | v0lZy | now that i've put it into that |
| 10:35.38 | v0lZy | i se |
| 10:35.39 | v0lZy | i see |
| 10:35.42 | v0lZy | ********Value********** |
| 10:35.47 | v0lZy | (value is what i put in) |
| 10:36.05 | v0lZy | but that means that exten => 015809316,n,Set(GLOBAL(OUTSIDECALLERID=0${CALLERID(number)})) is not doing anything |
| 10:36.37 | v0lZy | exten => 015809316,n,Set(GLOBAL(OUTSIDECALLERID=0${CALLERID(number)})) |
| 10:36.43 | EmleyMoor | Hmmm |
| 10:36.47 | v0lZy | why is it not overwritting OUTSIDECALLERID=value |
| 10:36.49 | v0lZy | ? |
| 10:37.33 | v0lZy | its as if its a constant? |
| 10:39.12 | EmleyMoor | Hmmm... that ought to work |
| 10:40.29 | v0lZy | <PROTECTED> |
| 10:40.30 | v0lZy | <PROTECTED> |
| 10:40.32 | v0lZy | <PROTECTED> |
| 10:41.17 | *** join/#asterisk asterisk_gurl (~asterisk_@119.234.0.34) |
| 10:41.23 | v0lZy | clearly this should be the culprit: "GLOBAL(OUTSIDECALLERID=00070551056)" |
| 10:41.36 | v0lZy | because |
| 10:41.44 | v0lZy | exten => 015809316,n,NoOP(********${OUTSIDECALLERID}********) |
| 10:42.12 | v0lZy | throws up value |
| 10:43.36 | v0lZy | how long can variable names be? |
| 10:44.40 | EmleyMoor | No idea - try shorter if you think it may help |
| 10:46.02 | v0lZy | can i use |
| 10:46.11 | v0lZy | Set(OUSTIDECALLERID=blahblabha,g) ? |
| 10:46.52 | EmleyMoor | No |
| 10:47.14 | EmleyMoor | Ah, hold on |
| 10:47.29 | EmleyMoor | Just spotted a possible error |
| 10:48.19 | EmleyMoor | Try Set(GLOBAL(OUTSIDECALLERID)=0${CALLERID(number)}) |
| 10:49.33 | v0lZy | <PROTECTED> |
| 10:49.34 | v0lZy | <PROTECTED> |
| 10:50.40 | v0lZy | it doesnt overwrite the value |
| 10:51.42 | EmleyMoor | Well, it should, unless I'm missing something |
| 10:52.33 | v0lZy | im still getting *********value********** instead of the number |
| 10:53.08 | v0lZy | http://www.the-asterisk-book.com/unstable/variablen.html |
| 10:53.13 | v0lZy | this link mentions a ,g flag |
| 10:53.17 | v0lZy | in Set |
| 10:53.24 | v0lZy | but i am not sure which asterisk version |
| 10:53.35 | EmleyMoor | Yes - that was used pre-1.4 |
| 10:54.15 | v0lZy | i tried commenting out the variable in the global section |
| 10:54.17 | v0lZy | no avail |
| 10:54.40 | v0lZy | funny thing |
| 10:54.44 | v0lZy | i still see 'value' there |
| 10:54.51 | v0lZy | even if i just uncomented it |
| 10:54.53 | v0lZy | erm |
| 10:54.54 | v0lZy | commented i mean |
| 10:55.10 | EmleyMoor | v0lZy: Has Asterisk been restarted? If not, it will still be there |
| 10:55.27 | v0lZy | just dialplan reload |
| 10:56.19 | EmleyMoor | They survive that |
| 10:56.44 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 10:56.44 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 10:57.12 | v0lZy | regardless |
| 10:57.17 | v0lZy | shouldnt i be able to overwrite it? |
| 10:57.37 | EmleyMoor | Yes, you should, unless I'm missing something. It's beyond me why you can't |
| 11:09.13 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a09e9.pool.mediaWays.net) |
| 11:10.40 | *** join/#asterisk shadebob (~shadebob@41.141.239.1) |
| 11:11.02 | *** join/#asterisk pyther (~pyther@unaffiliated/pyther) |
| 11:11.36 | shadebob | hi. I have a problem with my asterisk loading. When I start with asterisk -vvvvvc asterisk freeze on "IAX Ready and Listening" and stop to load other module … |
| 11:12.44 | *** join/#asterisk _zoom_ (~Eissa@196.1.219.122) |
| 11:15.39 | *** join/#asterisk _val_ (fbsd@178-84-212-245.dynamic.upc.nl) |
| 11:18.06 | _val_ | Hi everyone. I'm having the following issue when making a call on a foreign country. |
| 11:18.16 | _val_ | This is what I get. http://pastie.org/4329554 |
| 11:19.10 | WIMPy | Looks like your ITSP won't let you call spain. |
| 11:19.32 | *** join/#asterisk italorossi (~italoross@201.76.151.254) |
| 11:26.21 | _val_ | WIMPy: ITPS? |
| 11:26.34 | WIMPy | ~itsp |
| 11:26.34 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 11:26.53 | WIMPy | Whoever you sent that call to. |
| 11:30.48 | _val_ | WIMPy: Thank you. I'm going to find that out before I continue with this. I appriciate it! |
| 11:36.30 | tuxx- | hey guys, i got an asterisk 1.4 box here that crashes 4 times a day. The core show uptime tells me it has been running for 2 hours now. My idea is that somebody had to restart the asterisk proces in order to reset the uptime. Is the asterisk parent proces able to restart its childs and thus resetting the uptime? |
| 11:36.55 | tuxx- | Because none of the people that monitor the box did an asterisk restart as far as i can tell. |
| 11:37.07 | tuxx- | so i'm trying to figure out what happened exactly |
| 11:38.07 | WIMPy | It crashed and some init script started it again? |
| 11:38.22 | tuxx- | safe_asterisk isnt running |
| 11:38.31 | tuxx- | and the standard init.d script doesnt do that automatically right? :P |
| 11:38.37 | tuxx- | it has to be called by a user to do so |
| 11:38.46 | WIMPy | NFI |
| 11:38.54 | tuxx- | nfi? :? |
| 11:39.08 | tuxx- | ~nf |
| 11:39.09 | tuxx- | ~nfi |
| 11:39.09 | infobot | nfi is, like, No Fucking Idea |
| 11:39.12 | WIMPy | has never looked at the sullied scripts. |
| 11:39.12 | tuxx- | hehe |
| 11:39.17 | tuxx- | right |
| 11:39.37 | Roelt | crontab that restarts asterisk? |
| 11:39.52 | Roelt | depends on the linux distribution, scripts |
| 11:40.02 | tuxx- | nope, no crontab |
| 11:40.46 | tuxx- | its debian 6.0 with the standard asterisk 1.4 init.d script |
| 11:40.58 | WIMPy | Are you sure it's not restarted if it goes away? |
| 11:41.17 | tuxx- | nope, problem is that its a production environment, so i cant just kill asterisk :\ |
| 11:41.20 | Roelt | but solving the crashes would be a better use of your time than figuring out who/what restarted it.. :) |
| 11:41.25 | Roelt | +imho |
| 11:41.43 | WIMPy | Or just use a more recent version. |
| 11:41.53 | tuxx- | hehe |
| 11:42.15 | tuxx- | problem is |
| 11:42.20 | tuxx- | there are all kinds of ugly scripts running |
| 11:42.29 | tuxx- | the customer wrote himself |
| 11:42.34 | tuxx- | and the platform is gone within 2 months |
| 11:42.43 | tuxx- | meh |
| 11:43.33 | tuxx- | worst voip platform i ever dealt with |
| 11:43.34 | tuxx- | bah. |
| 11:43.34 | tuxx- | :D |
| 11:43.56 | WIMPy | Are there good ones? |
| 11:44.38 | tuxx- | i tend to think so |
| 11:45.44 | Roelt | friend of mine uses BroadWorks, he's happy with it |
| 11:49.40 | bdfoster | broadworks? |
| 11:51.00 | WIMPy | They have a nice marketing video. But I still have no idea, what they offer. |
| 11:52.57 | bdfoster | pretty much everything you could do on asterisk done in the cloud |
| 11:56.42 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-18-80.bois.qwest.net) |
| 11:57.54 | *** join/#asterisk irroot (~gregory@2002:294f:dde2:1:18d8:2584:a5f5:38e0) |
| 11:58.13 | irroot | I am back in civilization |
| 11:58.35 | irroot | stayed overnight took 7hrs there and back with the road works |
| 11:59.02 | WIMPy | Is civilization something good in your area? |
| 11:59.09 | irroot | prime tiger fishing country but no time to fish afternoon folks |
| 11:59.25 | irroot | WIMPy put pongola in google maps |
| 12:00.59 | *** join/#asterisk DynamicFail (~Administr@198.99.129.139) |
| 12:01.03 | DynamicFail | #join freepbx |
| 12:01.10 | WIMPy | Looks like anwhere here. |
| 12:01.30 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:01.56 | irroot | LOL @ DynamicFail |
| 12:02.26 | DynamicFail | <- appropriate name |
| 12:03.20 | irroot | WIMPy zoom out a bit :P its in the bush but hell palm trees 26c weather and green grass in winter is not a bad thing |
| 12:04.11 | WIMPy | needs a new home |
| 12:05.20 | WIMPy | LOL |
| 12:05.51 | WIMPy | Looks like you're about 10km south of a border, just like me :) |
| 12:06.29 | irroot | crossing those borders is a lotto :P |
| 12:07.03 | WIMPy | Panic! |
| 12:07.15 | WIMPy | Something stole my mouse pointer :-( |
| 12:07.15 | irroot | no one goes in that direction millions come in from there |
| 12:07.24 | WIMPy | blames google. |
| 12:07.26 | irroot | cat /dev/mouse |
| 12:07.48 | WIMPy | It's working, just invisible. |
| 12:09.22 | irroot | the mozimbique border is the bad one swaziland not bad but they have not had coup in 3 years so about due again |
| 12:09.44 | irroot | there king has like 20 wifes picks a new virgin every year |
| 12:13.31 | DynamicFail | Is there a roadmap plan for asterisk? |
| 12:13.42 | DynamicFail | nvm |
| 12:13.43 | DynamicFail | google |
| 12:14.53 | irroot | gtalk ?? |
| 12:15.24 | DynamicFail | nvm on the nvm (was saying used google to answer question, but then the page just says due to the nature of * we can't do a roadmap) |
| 12:15.27 | irroot | the problem with gtalk is its had a very unstable |
| 12:15.54 | irroot | what you looking for |
| 12:16.30 | DynamicFail | spec MLPP, fax support, decentralized development, hotline |
| 12:16.35 | irroot | there is a proposed features for each release and this depends often on if its a LTS or not |
| 12:17.18 | *** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh) |
| 12:17.37 | irroot | for example fax T38 gateway for sip was proposed for ast 10 and this needed to be added |
| 12:17.56 | irroot | in this case i took on the coding and did it with help from digium |
| 12:18.15 | irroot | if no one had taken up the proposal it would not be added |
| 12:19.02 | DynamicFail | how much would you say a feature request would be to develop for asterisk |
| 12:19.11 | DynamicFail | for example spec MLPP |
| 12:19.38 | irroot | its resource / requirement driven to a large extent |
| 12:19.38 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:20.15 | irroot | if there is enough support , understanding and willingness to add a feature it gets done |
| 12:20.15 | WIMPy | Loooking at Asterisks call handling capabilities, I guess that's not going to make too much sense beyond what you can do in the dialplan. |
| 12:20.26 | irroot | this is what the reviewboard process is for |
| 12:21.16 | irroot | JIRA if its a empty request |
| 12:21.22 | irroot | ie has no code |
| 12:23.51 | WIMPy | Dang. Do I really have to restart my whole desktop to get moy mouse pointer back? |
| 12:24.23 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:24.24 | *** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 12:25.06 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 12:26.31 | irroot | unplug plug back |
| 12:26.48 | irroot | open the mouse control panel change the color / icon |
| 12:26.51 | WIMPy | The moue is working. Just invisible :-( |
| 12:27.03 | bdfoster | windows? |
| 12:27.08 | WIMPy | If I could find such a thing... |
| 12:27.23 | bdfoster | WIMPy, you running windows? |
| 12:27.32 | WIMPy | I know it exists. I picked a bigger one, but I can't find out, where. |
| 12:27.41 | WIMPy | Nope. KDE |
| 12:27.55 | irroot | ctrl-alt-bksp |
| 12:28.02 | bdfoster | well, that's a step in the right direction |
| 12:28.04 | urvg4 | hi,how do I add multiple ip on one incoming sip trunk? is it host=ip1&ip2&ip3? |
| 12:28.21 | irroot | urvg4 use dynamic is possible ? |
| 12:28.22 | WIMPy | Yes, I guess I have to restart the whole thing :-( |
| 12:28.40 | bdfoster | urvg4, you don't. they're different trunks |
| 12:28.58 | WIMPy | urvg4: You need one peer per host. |
| 12:29.10 | urvg4 | irroot can't use dynamic as no password/username is provided |
| 12:29.15 | DynamicFail | Seems like the 2600hz project provides a distributed call management system |
| 12:29.38 | bdfoster | DynamicFail, it's a little more involved than that |
| 12:29.59 | urvg4 | ok thanks all |
| 12:30.25 | [TK]D-Fender | urvg4, you can't specify multiple hosts. So either make multiple peers or match by username |
| 12:30.34 | irroot | urvg4 ^^ wimpy use multiple definitions |
| 12:31.17 | irroot | one bad way is to use NAT but that is a bad idea |
| 12:31.54 | DynamicFail | what advantages does a 2600hz solution have over an asterisk solution running dundi? |
| 12:32.37 | bdfoster | no clue, dunno what dundi does |
| 12:32.53 | WIMPy | dundi is nice |
| 12:33.34 | bdfoster | i just use freeswithc |
| 12:33.37 | bdfoster | freeswitch |
| 12:34.11 | [TK]D-Fender | dundi is not for load balancing |
| 12:35.04 | irroot | i use macvlans with PPPoE accounts multiple then register multiple accounts one per link |
| 12:35.13 | WIMPy | But it can be used to locate peers on multiple Asterisks. |
| 12:35.20 | irroot | trick here is that they cant be sent to same IP |
| 12:35.29 | irroot | as the route cache interferes |
| 12:35.32 | bdfoster | remember that with the 2600hz stuff a ton of those functionalities that fs could handle on it's own are abstracted |
| 12:35.36 | bdfoster | heavily |
| 12:35.51 | irroot | it load balances but is crude |
| 12:36.32 | bdfoster | then it probably doesn't do HA |
| 12:36.41 | bdfoster | to the extent fs does anyway |
| 12:37.09 | bdfoster | i.e. fs1 crashes, call is continued on fs2 |
| 12:39.51 | WIMPy | WTF?!? I still have no mouse pointer after restarting X. |
| 12:40.10 | irroot | the cat chased /dev/mouse to /dev/null |
| 12:40.34 | WIMPy | will never use cat again |
| 12:41.33 | irroot | had a load problem on a system recently got the lady to see how i fix problem "killall -9 racoon" |
| 12:42.10 | irroot | naming utils after animals has some curious results |
| 12:42.39 | DynamicFail | bdfoster, I thought the situation you are describing is why fs was built |
| 12:42.53 | WIMPy | Using womens names for hostnames as well. |
| 12:43.08 | irroot | hehehe finger ..... |
| 12:43.50 | irroot | ssh root@..... |
| 12:46.45 | DynamicFail | [TK]D-Fender, I'm confused does 2600hz allow you to create your own cloud, or do they just require you to use their own servers? I'm looking for something that would allow call managers to come up and down frequently (in and out of the cloud) and still maintin voip communications |
| 12:47.09 | [TK]D-Fender | DynamicFail, I haven't touched FS at all yet |
| 12:49.19 | bdfoster | DynamicFail, it can be done both ways |
| 12:49.48 | bdfoster | yes, fs was build for ha/load balancing |
| 12:50.28 | bdfoster | but when you're talking about something as complex as a business class pbx with things like conferences, parking lots, etc. there is a need for abstraction |
| 12:50.36 | *** join/#asterisk zamba (marius@flage.org) |
| 12:55.23 | *** join/#asterisk cyborg-one (1000@79-140-2-8.broadband.tenet.odessa.ua) |
| 12:57.21 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-dloiohbyiplouoss) |
| 12:57.21 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:58.20 | DynamicFail | Seems like Asterisk SCF is going to do load balancing |
| 12:58.54 | *** join/#asterisk shadebob (~shadebob@41.248.242.85) |
| 13:00.44 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 13:14.56 | *** part/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 13:20.21 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 13:22.49 | bdfoster | off to work bbl |
| 13:24.37 | DynamicFail | seems like freeswitch is still being actively developed and may be larger than asterisk now? |
| 13:25.19 | bdfoster | ...i wouldn't say that |
| 13:25.29 | bdfoster | it's for sure being actively developed |
| 13:25.40 | bdfoster | with sometimes hundreds of commits daily |
| 13:25.58 | bdfoster | but bigger than asterisk, that's probably not the right choice of words... |
| 13:29.01 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-taavvvaoqlgyatms) |
| 13:29.06 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:30.14 | *** join/#asterisk serafie (~erin@nat/digium/x-vrvnntnoynvhcdsa) |
| 13:34.44 | *** join/#asterisk jkroon (~jkroon@dsl-244-38-32.telkomadsl.co.za) |
| 13:35.41 | jkroon | hi guys, i just submitted a case to digium, and was hoping I can get hold of someone that will take a decent look at it, it's quite an annoying g729 license issue that has cost me quite a bit of money already, and i'd like to get a long-term solution. |
| 13:36.11 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-yfutkxaklormdzsc) |
| 13:37.56 | DynamicFail | Do you think any asterisk experts would be interested in doing some youtube "interviews" on various topics |
| 13:38.40 | DynamicFail | I was going to start a site where I put clips from experts of various programs where they describe high level info on it |
| 13:39.05 | jkroon | DynamicFail, i'm sure you'll be able to convince a few peeps. |
| 13:39.17 | Katty | :> |
| 13:39.56 | DynamicFail | nice, I'm still working on the format etc on how I want to do it, but I thought it would be really cool |
| 13:40.34 | DynamicFail | to be able to see and hear a lot of these experts explain things. |
| 13:40.36 | WIMPy | jkroon: #asterisk-bugs |
| 13:41.31 | newtonr | jkroon: for your Digium case, you would need to contact Digium support for an update on it |
| 13:41.43 | newtonr | jkroon: did you submit a bug report via JIRA, or did you submit a case to support? |
| 13:41.47 | jkroon | busy calling now ... |
| 13:41.55 | jkroon | newtonr, submitted a case to support. |
| 13:43.24 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:43.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:43.27 | DynamicFail | http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/ <<== describes asterisk being used for load-balancing and "fully-distributed" with failover |
| 13:43.33 | DynamicFail | now I'm real confused |
| 13:45.05 | wonderworld | will registered sip clients stay registered, if their credentials are removed from sip.conf? |
| 13:45.34 | WIMPy | Until you 'sip reload'. |
| 13:45.53 | WIMPy | That is, do you only remove the credentials or the peer? |
| 13:46.17 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:46.17 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:46.31 | [TK]D-Fender | wonderworld, calls will fail without an entry to auth them |
| 13:46.53 | wonderworld | will already established calls be dropped? |
| 13:47.41 | WIMPy | But even if they cant auth, they should still stay registered until the registration times out. |
| 13:48.56 | wonderworld | thank you guys |
| 13:52.28 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 13:53.46 | *** join/#asterisk shadebob (~shadebob@41.248.242.85) |
| 13:53.51 | shadebob | Hi, my asterisk isn't fully booted. When I start with asterisk -vvvvvvc he stop on IAX Ready and Listening but he doesn't give me the CLI |
| 13:53.54 | wonderworld | hmm, i tested it. calls are possible, even if credentials have changed and "sip reload" has been used. |
| 13:55.17 | WIMPy | wonderworld: But only in one direction. |
| 13:55.27 | WIMPy | Unless you let everybody in. |
| 13:55.56 | *** join/#asterisk bchia (~Adium@nat/digium/x-gsvgesadvjkzwaom) |
| 14:00.48 | [TK]D-Fender | However being registered without a peer doesn't get you anywhere wither |
| 14:00.55 | [TK]D-Fender | the incoming call won't get matched either way |
| 14:01.00 | [TK]D-Fender | the device will still think it' |
| 14:01.04 | [TK]D-Fender | s OK perhaps |
| 14:01.06 | [TK]D-Fender | for a bit |
| 14:01.17 | [TK]D-Fender | however I expect any re-reg attempt to fail anyway |
| 14:01.29 | [TK]D-Fender | and it would probably look like it's falling off the map when it does |
| 14:01.51 | WIMPy | thinks the same |
| 14:02.34 | WIMPy | Or maybe just removing credentials won't work, but you have to set them to something different. |
| 14:03.05 | WIMPy | had several occasions where 'sip reload' didn't produce the expected result, but reloading chan_sip did. |
| 14:13.36 | *** join/#asterisk sustav (~vpp@nat/digium/x-sqzoordotenqhlda) |
| 14:27.17 | *** join/#asterisk eject_ck (~eject@109.86.138.135) |
| 14:30.05 | schmidts | WIMPy database drop SIP/Registry also helps with sip reload ;) |
| 14:30.17 | *** join/#asterisk kaspar (~kaspar@p50811DF9.dip.t-dialin.net) |
| 14:30.21 | eject_ck | Hi all. I have problem with chan_ooh323 - provider says that they receive empty callerid, where I've set it in dialplan using exten => _X.,1,Set(CALLERID(all)=+<myphonethere>) |
| 14:31.02 | eject_ck | I've set callerid in ooh323.conf as well with same result |
| 14:31.04 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:31.39 | WIMPy | schmidts: Yes, that will help. |
| 14:33.30 | kaspar | Hi all. I've a question regarding the CHANNEL function. I can get chan_sip's RTP stats on an incoming SIP context with "exten => h,1,Set(CDR(userfield)=${CHANNEL(rtpqos,audio,all)})" just fine. But how can I access these stats when coming from a local (non-sip) channel which Dial()s out via SIP? |
| 14:33.35 | eject_ck | Should it work with ooh323 ? |
| 14:34.17 | WIMPy | eject_ck: It should work with any channel. |
| 14:35.09 | *** join/#asterisk saysocomm (~dotcomm@74-92-251-226-Tallahassee.hfc.comcastbusiness.net) |
| 14:35.11 | kaspar | I basically want to log all RTP stats from all incoming and outgoing chan_sip calls... |
| 14:35.39 | DynamicFail | Does Dundi use DNS SRV? |
| 14:36.07 | WIMPy | DynamicFail: For what? |
| 14:36.26 | DynamicFail | communicating between servers |
| 14:36.29 | DynamicFail | to keep them updated |
| 14:38.40 | *** join/#asterisk ideaman (~ihaveapla@173-10-29-218-BusName-utah.ut.hfc.comcastbusiness.net) |
| 14:44.35 | ideaman | Anyone: If I'm compiling dahdi 2.6.1 and the svn of 1.8, I compiled Dahdi first, then asterisk. But, does it matter at which point I install Libpri? |
| 14:45.11 | [TK]D-Fender | Should be first |
| 14:45.15 | WIMPy | Before Asterisk. |
| 14:45.38 | ideaman | k |
| 14:45.44 | ideaman | I'll go try that |
| 14:45.47 | ideaman | thx |
| 14:53.28 | *** join/#asterisk mrchicken (be93997b@gateway/web/freenode/ip.190.147.153.123) |
| 14:53.33 | mrchicken | 'ello |
| 14:54.08 | Katty | bergawwwk! |
| 14:54.09 | shadebob | Re, My asterisk isn't full loaded (asterisk -vvvvvc doesn't give cli) because my astdb is corrupted .... |
| 14:54.59 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:55.05 | mrchicken | shadebob: I'd save the configs and reinstall :P |
| 14:55.21 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 14:56.06 | shadebob | everything it's ok now … I wrote this message just for info. |
| 14:57.27 | WIMPy | shadebob: How did you break it? |
| 14:57.27 | mrchicken | Anybody has any idea why asterisk wont play a custom sound file I just uploaded to the sounds dir? |
| 14:57.46 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 14:57.46 | mrchicken | Its got the right permissions... |
| 14:58.00 | WIMPy | And the right format? |
| 14:58.03 | shadebob | I don't known… my customer use Elastix … |
| 14:58.15 | WIMPy | Without any extras Asterisk doesn;t understand? |
| 14:58.16 | shadebob | so astdb is VERY HUGE. |
| 14:58.42 | shadebob | more than 500 extensions in server |
| 14:59.16 | drmessano | 500 extensions on an Elastix box? |
| 14:59.35 | shadebob | symptom : asterisk doesn't make a complete reload. An asterisk -vvvvvvr load all module but doesn't give me the CLI |
| 14:59.41 | WIMPy | Does it strecht that far? |
| 15:00.12 | shadebob | solution : unload all modules …. load one per one …. and catch sqlite3 corrupted db …. |
| 15:00.16 | drmessano | I know it's never been recommended to put more than 150 extensions in the FreePBX database. Unless something changed recently |
| 15:00.29 | WIMPy | Has someone made span-dex already, BTW? |
| 15:00.36 | shadebob | drmessano : not my choice :( |
| 15:00.43 | drmessano | Choice has nothing to do with it |
| 15:00.45 | kaspar | hey all, I solved my problem by adding "e" option to my Dial() targets. That way the h extensions get dialled for both channels. |
| 15:00.59 | vastina | hell we can make a bilingual one called... "his-span-dex" |
| 15:01.09 | drmessano | What I am telling you is that what you have is not a supported configuration... and that throws a lot of things out the window |
| 15:01.17 | WIMPy | nice |
| 15:02.27 | drmessano | Of course.. the other issue is, Elastix uses an older FreePBX.. and while it may scale better now, it definitely didn't when 2.8 was new |
| 15:03.16 | shadebob | drmessano : It seem works well for 2 years now. I just know the customer. |
| 15:03.49 | drmessano | The astbd corruption is generally an easy fix |
| 15:03.57 | drmessano | astdb |
| 15:05.10 | drmessano | If you google for freepbx and resetall, there is a string you can add to the URL in the Web UI that will purge the astdb and rebuild it from the FreePBX DB |
| 15:06.23 | shadebob | Client have incremential backup … so no problem. Just wasting my time to find the problem :( |
| 15:08.34 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 15:09.02 | mrchicken | I have a TDM400 that wont detect a hangup... Ive set busydetect=yes busycount=6 and hanguponpolarityswitch=yes. What else am I missing? |
| 15:09.04 | drmessano | astdb corruption happens |
| 15:10.44 | shadebob | mrchicken : busypattern=500,500 ? |
| 15:10.59 | shadebob | and progzone ? |
| 15:11.01 | mrchicken | isnt that the default? |
| 15:11.03 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:11.43 | shadebob | it depends on your country |
| 15:12.34 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-18.nwrknj.fios.verizon.net) |
| 15:15.26 | eject_ck | Checked my call with wireshark, I see that from is empty :( |
| 15:17.36 | WIMPy | eject_ck: That's most certainly not the intention. |
| 15:18.54 | *** join/#asterisk LedZeplin (jbearer@shell.atvnc.com) |
| 15:19.45 | eject_ck | WIMPy: exten => _X.,1,Set(CALLERID(all)=+<mynumber>) replaced by exten => _X.,1,Set(CALLERID(all)=0000000000002) |
| 15:19.49 | eject_ck | and it works now! |
| 15:20.07 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:21.17 | WIMPy | So it's the +? |
| 15:22.39 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 15:24.20 | LedZeplin | We're switching our PRI over to a new provider and I'm not sure what I need to change based on the information the new provider provided. I think some of these values are redundant when configuring asterisk. Here is the details of what they say, and what I have currently: http://tinypaste.com/bbb13116 |
| 15:27.41 | WIMPy | Why do you expect to change more than a cable? |
| 15:27.48 | WIMPy | doesn;t see anything in that pb. |
| 15:28.34 | LedZeplin | WIMPy: I am just doing due dilligance to make sure things go right. |
| 15:29.27 | LedZeplin | thanks for the consult. I was unsure since they said they use a DMS100 and that is an option instead of national for switchtype. |
| 15:30.22 | WIMPy | And what do they use? |
| 15:30.22 | LedZeplin | and I couldn't find how pri_cpe signaling related to either NI2 D channel or the B8ZS - ESF signaling mentioned by the new provider. |
| 15:30.55 | WIMPy | Different levels and different config files. |
| 15:31.05 | WIMPy | got to go |
| 15:31.47 | LedZeplin | hrm, I've been grepping, what is the other config file? I'm looking at the dahdi-channels.conf |
| 15:32.10 | *** join/#asterisk TomCat2 (~TomCat@static-72-77-202-50.tampfl.fios.verizon.net) |
| 15:35.07 | [TK]D-Fender | switchtype=dms100 <--------- |
| 15:35.15 | [TK]D-Fender | not "national" |
| 15:35.42 | [TK]D-Fender | and pri_cpe is right for you being the CPE side |
| 15:41.31 | LedZeplin | [TK]D-Fender: ok thanks for the signaling doublecheck since I couldn't find anything that linked "pri_cpe" to the standards they provided. |
| 15:42.10 | [TK]D-Fender | ledthat jsut says that you are the Customer |
| 15:42.16 | LedZeplin | Now I just need a 3rd person to break the tie! however i'm leaning towards switchtype=dms100 to be the correct config. |
| 15:42.55 | LedZeplin | Where are the configs, or what configs cover the B8ZS - ESF and the NI2 D channel signaling? |
| 15:47.56 | *** join/#asterisk doolittlework (~doolittle@41-134-22-14.dsl.mweb.co.za) |
| 15:49.11 | [TK]D-Fender | NI2 = national 2. Aka "not applicable" as you're on DMS100 |
| 15:49.14 | LedZeplin | duh I found it in /etc/dahdi/ |
| 15:49.19 | [TK]D-Fender | the rest of teh framing & coding is STANDARD for T!. |
| 15:49.23 | [TK]D-Fender | T1 |
| 15:49.29 | LedZeplin | thanks [TK]D-Fender |
| 15:50.47 | *** join/#asterisk Hotwired (~ed@unaffiliated/hotwire) |
| 15:56.46 | Katty | i'm having a blonde moment. |
| 15:56.58 | Katty | where are the recorded calls stored >.< |
| 15:57.16 | Katty | for some reason i thought it was var spool asterisk monitor |
| 15:57.29 | Katty | OHWAIT, totally a blonde moment. |
| 15:57.48 | Katty | thwaps self with cluebat(tm) |
| 15:58.06 | drmessano | Thank you for respecting trademarks |
| 15:58.30 | [TK]D-Fender | ClueBat <- required caps |
| 16:01.31 | drmessano | I hate Windows |
| 16:01.43 | drmessano | Can a ClueBat be used to break Windows? |
| 16:02.40 | drmessano | I would have named it Doors and had it play "The End" when it booted |
| 16:10.20 | *** join/#asterisk mrchicken (be93997b@gateway/web/freenode/ip.190.147.153.123) |
| 16:11.46 | mrchicken | Hi. when I do Background(custom/fileName) I get a message that reads File no-valid-responce-transfering does not exist in any format |
| 16:12.08 | mrchicken | but the file is the sounds/custom dir!! |
| 16:13.03 | *** join/#asterisk tully` (Tully@74.195.67.216) |
| 16:13.38 | tully` | Does anyone have any experience with sipp? I'm trying to load test an asterisk box and could use some help. |
| 16:15.11 | pyther24 | tully`: that's like asking if anyone has any linux experience :-/ |
| 16:15.34 | tully` | sipp as in the stress testing tool, not the protocol |
| 16:16.23 | drmessano | I guess reading experience is even less common |
| 16:16.46 | v0lZy | hey guys |
| 16:16.50 | v0lZy | anyone know for a soltion |
| 16:16.59 | v0lZy | to get caller id when doing call pickup? |
| 16:17.33 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:20.35 | Qwell | mrchicken: So, you are trying to play a file named fileName, and it fails with a different filename? |
| 16:21.02 | mrchicken | nope, it fails with the same name |
| 16:21.13 | Qwell | show me. |
| 16:21.24 | mrchicken | I do Background(custom/bienvenidoscali) |
| 16:22.04 | Qwell | and the error? |
| 16:22.16 | mrchicken | lemme pastebin it |
| 16:22.30 | Qwell | ls -la /var/lib/asterisk/sounds/custom/bienvenidoscali.* |
| 16:22.59 | Katty | i'll ls you in a minute. |
| 16:23.18 | Qwell | Katty: PG13! |
| 16:23.22 | mrchicken | http://pastebin.ca/2174604 |
| 16:23.35 | Katty | Qwell: i am totally PG13. |
| 16:23.47 | mrchicken | Qwell: nope, this is an R space |
| 16:24.36 | Qwell | mrchicken: and the output of the ls? |
| 16:25.11 | mrchicken | root@phoneApp:/var/lib/asterisk/sounds/custom# ls -l total 4 -rw-r--r-- 1 asterisk asterisk 3399 jul 24 14:09 bienvenidoscali.gsm root@phoneApp:/var/lib/asterisk/sounds/custom# |
| 16:25.23 | tully` | What I'm trying to do is see how many concurrent calls my asterisk machine can handle, but I need it to go through the dialplan and simulate an actual user. I just need it to call, wait a few seconds, send a DTMF of 1, then wait some more and hangup. |
| 16:25.24 | leifmadsen | there is now ulaw file |
| 16:25.32 | leifmadsen | s/now/no |
| 16:26.02 | mrchicken | but I just did background(custom/bienvenidoscali) |
| 16:26.07 | mrchicken | should I append the .gsm??? |
| 16:26.11 | leifmadsen | no |
| 16:26.32 | mrchicken | so? |
| 16:26.35 | leifmadsen | do you have a language defined? perhaps it needs to be a custom directory under /var/lib/asterisk/sounds/<lang>/ |
| 16:26.54 | mrchicken | hrmmm maybe its that |
| 16:26.57 | mrchicken | lemme check |
| 16:27.58 | mrchicken | not that I am aware of |
| 16:28.18 | mrchicken | or hold on... that would be in the config for the dahdi channels, right? |
| 16:29.39 | mrchicken | I dont see anything |
| 16:29.58 | mrchicken | how do I check the language thing? |
| 16:31.27 | mrchicken | nope |
| 16:31.30 | mrchicken | no language |
| 16:31.41 | mrchicken | just checked the dahdi channels and no language at all |
| 16:32.06 | leifmadsen | maybe the file isn't of the correct format for asterisk to play |
| 16:32.10 | leifmadsen | or there is a permissions issue |
| 16:32.15 | leifmadsen | otherwise I have no further ideas |
| 16:34.34 | [TK]D-Fender | I see no proof of the file's actual format, or that the folders in the chain are right, or that that is eevn where * is looking for them. No complete CLI output, etc... |
| 16:34.47 | [TK]D-Fender | I trust very little of what little I see |
| 16:36.04 | paulc | Hey leifmadsen :) - Remember last week I was blathering on about CURL on one channel blocking CURL execution on another channel? |
| 16:36.13 | leifmadsen | I do! |
| 16:37.33 | paulc | Known issue apparently: https://issues.asterisk.org/jira/browse/ASTERISK-18708 |
| 16:37.43 | leifmadsen | huh! |
| 16:38.08 | paulc | For the majority of my stuff, not an issue, cos it's always hitting localhost for the web services/PHP/MySQL magic... but when we hit an external vendor, and they're having "issues", it makes everything else grind to a halt |
| 16:38.37 | leifmadsen | eep |
| 16:39.17 | paulc | Built a test case with 2 scripts, one with a sleep(10) and one with a sleep(2) and it behaves as reported (ie wrongly! blocking = BAD!) - The attached patch seems to work in my test environment, but I haven't made the change in production.. slightly hesitant to have to patch the regular releases :-s |
| 16:40.58 | paulc | Figured I'd give you the heads up cos I know you do a bunch of CURL-y stuff too.. and perhaps have a bit more sway with "the powers that be" to delve deeper into the issue - it's potentially a bit of a show stopper :( |
| 16:41.25 | leifmadsen | paulc: that sucks :( ya I don't really do a lot of asterisk stuff anymore to be honest :( |
| 16:41.41 | WIMPy | v0lZy: Doesn't it do that now? What Version? |
| 16:41.49 | paulc | ah no? moved on? What's the flavour du jour now? |
| 16:41.56 | v0lZy | 1.8 |
| 16:42.06 | v0lZy | if i do call pickup, i dont get anything |
| 16:42.13 | v0lZy | unless i should use trustrpid? |
| 16:43.22 | WIMPy | Sendrpid, surely |
| 16:45.19 | WIMPy | should pry to do pickup again... |
| 16:46.14 | *** join/#asterisk karl370 (~karl370@12.139.131.2) |
| 16:46.53 | v0lZy | senrpid=pai |
| 16:46.54 | v0lZy | i have that |
| 16:47.02 | v0lZy | but trustrpid i dont have |
| 16:47.10 | v0lZy | beause then i get unknowns from certain providers |
| 16:50.34 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 16:50.43 | WIMPy | wonders if it's possible to do pickup without using PICKUPMARK everywhere. |
| 16:55.27 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 16:56.32 | v0lZy | WIMPy: the problem i guess is modifying the header once the whole thing is already established |
| 16:56.42 | v0lZy | since theuser dials *8extension |
| 16:56.52 | v0lZy | thats what the users sees on his phone |
| 16:57.05 | WIMPy | That happens all the time with CONNECTEDLINE etc. |
| 16:57.21 | v0lZy | what do u mean? |
| 16:57.52 | v0lZy | and how does this integrate with BLFs... ? |
| 16:58.10 | WIMPy | There's no interaction with BLF. |
| 16:58.16 | v0lZy | k |
| 16:58.25 | v0lZy | then.. any way to get it to update the scren after i do *8XX |
| 16:58.27 | WIMPy | It just updates yur display. |
| 16:58.27 | v0lZy | ? |
| 16:58.41 | v0lZy | what parameters are needed? |
| 16:58.45 | v0lZy | and in which conf file |
| 16:58.47 | v0lZy | for me, it doesnt do that |
| 16:58.50 | v0lZy | i just see what i dialed |
| 16:58.57 | Qwell | Does your phone support it? |
| 16:59.06 | v0lZy | aastra phone should |
| 16:59.12 | v0lZy | so should the draytek i think |
| 16:59.22 | WIMPy | The pickup application would have to do that in case of a pickup. |
| 16:59.54 | *** join/#asterisk ruben23 (~John_Carl@121.97.111.142) |
| 16:59.59 | v0lZy | i think its mostly pickupmark everywhere |
| 17:00.11 | v0lZy | in my dialplan which im trying to decypher and augument |
| 17:00.24 | WIMPy | That's wht I just tried. But I don;t seem to get a match. |
| 17:00.25 | v0lZy | thing is, if i ring XX and try to pick that up from YY |
| 17:00.37 | v0lZy | i see XX on the screen of YY, not the external number |
| 17:00.39 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 17:01.22 | WIMPy | I tell you if I get an update if you tell me how to make pickup work. |
| 17:02.04 | v0lZy | make sure to Set(__PICKUPMARK@<number>) |
| 17:02.05 | ruben23 | hi guys any have guide how to setup voicemail on asterisk email directly to gmail..? when someone comes in and created..? any ide aguys |
| 17:02.09 | v0lZy | __ |
| 17:02.15 | WIMPy | I just always Set(PICKUPMARK=1) and then do a Pickup(1@PICKUPMARK). Shouldn;t that always work? |
| 17:02.22 | v0lZy | key here... so u can use it down the road |
| 17:02.32 | v0lZy | erm |
| 17:02.34 | v0lZy | try |
| 17:02.36 | v0lZy | __PICKUPMARK |
| 17:02.42 | v0lZy | instead of PICKUPMARK=1 |
| 17:02.44 | v0lZy | try |
| 17:02.46 | WIMPy | Hmm. Is it the outgoing channels that are picked? |
| 17:02.51 | v0lZy | __PICKUPMARK=1 |
| 17:02.55 | WIMPy | Could make sense for the other way, I guess. |
| 17:03.22 | v0lZy | if u don tuse the __ |
| 17:03.36 | WIMPy | Yes, got it. |
| 17:03.43 | v0lZy | works? |
| 17:04.10 | WIMPy | Yes, but I wonder what I did next. |
| 17:05.06 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 17:06.29 | WIMPy | I can't pickup from SIP. |
| 17:06.39 | WIMPy | wonders what means again. |
| 17:07.34 | v0lZy | bhmm? |
| 17:08.22 | WIMPy | Nothing on *CLI except for Using SIP RTPP TOS bits ans CoS mark. Ans a 404 forbidden on the phone. |
| 17:08.29 | WIMPy | Works with the other phones however. |
| 17:09.46 | [TK]D-Fender | 404 = not found, not "forbidden" |
| 17:09.54 | v0lZy | 403 = forbidden |
| 17:09.58 | v0lZy | at leaast in http :D |
| 17:10.15 | v0lZy | WIMPy: what u mean cant pickup from sip? |
| 17:10.22 | WIMPy | Sorry. It just says forbidden. |
| 17:11.38 | v0lZy | u set pickupmarks |
| 17:11.46 | v0lZy | but u cant pickup using them? |
| 17:12.06 | v0lZy | pickup groups? |
| 17:12.39 | WIMPy | 100 Trying; 403 Forbidden. |
| 17:13.03 | WIMPy | Groups shouldn't matter with PICKUPMARK, should they? |
| 17:13.15 | WIMPy | I can pickup from the old phones. |
| 17:13.29 | v0lZy | dont know but |
| 17:13.38 | v0lZy | sip.conf add the phones to the same pickupgroup |
| 17:13.39 | v0lZy | see if that helps |
| 17:13.49 | v0lZy | if nothing, u rule it out |
| 17:15.07 | WIMPy | Maybe because the sip channel has gone? |
| 17:15.10 | WIMPy | checks. |
| 17:15.29 | WIMPy | nope that's not the issue. |
| 17:15.58 | WIMPy | goes to pick a 2nd sip phone... |
| 17:16.52 | *** join/#asterisk Champi (Champi@rootshell.fr) |
| 17:21.08 | WIMPy | It looks like I can't dial that extension from SIP. But I have NFI, why. |
| 17:21.46 | WIMPy | I put a verbose in front and that isn't hit. The context, however is the same. |
| 17:23.00 | WIMPy | I looks liek it doesn't even try to enter the dialplan. |
| 17:23.34 | v0lZy | hammer? :D |
| 17:25.00 | WIMPy | What was the definition of RAM? |
| 17:25.24 | WIMPy | RAM: What a programmer would like to to to the computer when still working at midnight. |
| 17:26.03 | WIMPy | Any idea how to debug this? |
| 17:26.33 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 17:26.57 | v0lZy | place noops where u think the dialplan is going |
| 17:27.01 | v0lZy | see if its going there? |
| 17:27.05 | WIMPy | The dialling method seems unrelated as well. |
| 17:27.19 | WIMPy | Already did. It's not going there. |
| 17:27.33 | v0lZy | found out where its going? |
| 17:28.05 | WIMPy | Nowhere. verbose/debug doesn't give anything. |
| 17:28.31 | WIMPy | In SIP debug I see 100 Trying immediately followed by 403 Forbidden. |
| 17:28.33 | v0lZy | and the call crashes? |
| 17:28.39 | WIMPy | But the phone already told me that. |
| 17:29.00 | v0lZy | reboot? :D |
| 17:29.17 | WIMPy | Windows repair on Asterisk. |
| 17:29.26 | WIMPy | Well, it can't hurt. |
| 17:30.03 | WIMPy | Nope. No change. |
| 17:30.16 | v0lZy | have u tried bribing it? :D |
| 17:30.25 | v0lZy | seriously |
| 17:30.34 | WIMPy | With overvoltage? |
| 17:30.38 | v0lZy | there should be a bribe clause in computing |
| 17:30.46 | WIMPy | I'd often like to. |
| 17:31.12 | v0lZy | GiveIf (this evalueates true; more components) |
| 17:31.24 | v0lZy | and also.. threat! |
| 17:31.38 | v0lZy | TakeIf (this evaluates false; less components, bwahahahaha) |
| 17:31.53 | v0lZy | work || uninstall :D |
| 17:31.56 | [TK]D-Fender | WIMPy, if it 403's that call then it should refuse ALL calls |
| 17:32.31 | WIMPy | It's only the pickup extension. |
| 17:32.43 | [TK]D-Fender | show both |
| 17:32.47 | [TK]D-Fender | and all the backup |
| 17:33.34 | WIMPy | Ha. Could it be "features" that interfere? |
| 17:33.51 | v0lZy | yes |
| 17:33.59 | v0lZy | i think theres an option to not allow pickups |
| 17:34.31 | Katty | which of the polycom config files is the one that handles the volume |
| 17:34.36 | WIMPy | No, it looks lile I had the same pickupexten there (probably from the dark ages or so). |
| 17:34.37 | Katty | the volume of the call |
| 17:34.54 | v0lZy | or actually theres just pickupexten=*8 |
| 17:35.05 | WIMPy | That was the evil one. |
| 17:35.15 | WIMPy | Interesting that it only interferes with SIP. |
| 17:36.00 | WIMPy | But now that it finally works I can tell you that the phone displays the number of the caller after pickup. |
| 17:36.26 | WIMPy | . o O ( This feature shit is EEEEVIL ) |
| 17:36.34 | v0lZy | and what are your other settings? |
| 17:36.50 | v0lZy | are you trusting rpid? |
| 17:36.50 | WIMPy | Which ones? |
| 17:37.10 | WIMPy | no |
| 17:37.27 | v0lZy | hm |
| 17:37.31 | WIMPy | rpid_update=yes looks important |
| 17:38.15 | v0lZy | i have |
| 17:38.20 | v0lZy | ;trustrpid=yes |
| 17:38.22 | v0lZy | sendrpid=pai |
| 17:38.23 | v0lZy | rpid_update=yes |
| 17:38.33 | v0lZy | not trusting rpid as its comented out |
| 17:38.39 | v0lZy | doing rpid_update... |
| 17:38.44 | v0lZy | sendripid=pai... |
| 17:38.48 | WIMPy | same here except for an explicit trustrpid=no |
| 17:38.51 | v0lZy | and when im transfering |
| 17:39.10 | v0lZy | i got it working, even with attended transfers thanks to doolittlework and his magic |
| 17:39.35 | v0lZy | when transfering attended or unattended i send numbers.... |
| 17:39.40 | v0lZy | but with pickup, i just see what im entering |
| 17:39.50 | v0lZy | do the phones need a setting to allow this update_rpid? |
| 17:39.57 | v0lZy | rpid_update i mean. |
| 17:40.17 | WIMPy | Well, if it works for transfers, it looks like everything is setup right. |
| 17:40.36 | WIMPy | Maybe Pickup() didn;t have the feature for too long? |
| 17:41.20 | v0lZy | no actually |
| 17:41.22 | v0lZy | for transfersž |
| 17:41.24 | v0lZy | its like this |
| 17:41.29 | v0lZy | blind transfers are not a problem |
| 17:41.45 | v0lZy | that just required sendrpid=pai |
| 17:42.03 | v0lZy | attended transfers requierd heavy modification of internal dialplan and incoming provider dialplan |
| 17:42.40 | ruben23 | hi guys any have guide how to setup voicemail on asterisk email directly to gmail..? when someone comes in and created..? any ide aguys |
| 17:42.42 | v0lZy | pickup stuff. ... now that just aint working |
| 17:43.15 | WIMPy | ruben23: Have you take a look in to the voicemail.conf file already? |
| 17:43.45 | v0lZy | we even got it to show two numbers when transfering calls attendedly. |
| 17:44.24 | WIMPy | Hmm. Interesting. When I start an attended transfer I get "s" as caller number. But after the transfer completes, the caller number is shown. |
| 17:44.38 | *** join/#asterisk italorossi (~italoross@189.124.200.92) |
| 17:45.19 | WIMPy | Do feature transfers use ${EXTEN} as callerid? |
| 17:45.20 | v0lZy | i get |
| 17:45.31 | v0lZy | first the caller number and under that, the person thats doing the transfering |
| 17:45.54 | v0lZy | in the solution ih ave here |
| 17:46.00 | WIMPy | That must be blind transfer then. |
| 17:46.22 | ruben23 | WIMPy:yes..but how do i forward it to email , i have default sendmail client |
| 17:46.22 | v0lZy | we set the DID of the phone thats called to the callers number |
| 17:46.44 | v0lZy | then when the transfer hits the transfer-to extensions dialplan under internal |
| 17:46.50 | WIMPy | ruben23: Just specify the destination address. |
| 17:47.00 | WIMPy | v0lZy: SIP transfers look better. |
| 17:47.02 | v0lZy | we fetch callerid and based on that callerid go lookup what the original callers number is in the database |
| 17:47.13 | [TK]D-Fender | ruben23, point the Vm entry to that address. The End |
| 17:47.23 | v0lZy | then inject tha tinto callerid before doing the dial |
| 17:47.46 | WIMPy | v0lZy: Seems to work quite well out of the box (by now). |
| 17:47.55 | v0lZy | WIMPy: i how do i enable those? |
| 17:48.17 | WIMPy | But where can I set the callerid for feature style attended transfers? |
| 17:49.03 | WIMPy | would live to completely disable that feature shit if it wasn't for the odd shitty phone or channel. |
| 17:49.08 | v0lZy | dont know what u mean, are there two ways of doing an attended transfer? |
| 17:49.19 | WIMPy | Yes |
| 17:49.31 | v0lZy | uh, how? |
| 17:49.48 | WIMPy | Via Asaterisks "featues" or via soem channel specific way. |
| 17:50.09 | v0lZy | asterisk features ... like doing ##something and **something? |
| 17:50.17 | WIMPy | That is the DTMF stype vs using some dedicated button. |
| 17:50.22 | WIMPy | Yes |
| 17:50.38 | v0lZy | thats kinda ugly |
| 17:50.56 | v0lZy | i mean dont get me wrong, its probably better |
| 17:50.58 | WIMPy | d/kinda/ |
| 17:51.04 | v0lZy | but my phone has blf |
| 17:51.11 | v0lZy | and users want to use those |
| 17:51.31 | WIMPy | Yes, that's definitely the better way. |
| 17:51.42 | v0lZy | and do blind transfers or attended transferes based on what they do with the headset |
| 17:52.02 | v0lZy | but that way of doing attended transfer doesnt work out of the box in my case |
| 17:52.05 | WIMPy | That's the way it should be. |
| 17:52.25 | v0lZy | the receiver doesnt see the original callers number, but the person who they got the transfer from's number. |
| 17:52.41 | v0lZy | blindtransfer is ok |
| 17:52.53 | WIMPy | Until the transfer is completed. |
| 17:53.04 | WIMPy | Which makes perfect sense. |
| 17:53.07 | v0lZy | no |
| 17:53.09 | v0lZy | even after its completed |
| 17:53.19 | v0lZy | when the transfering guy gets off the line |
| 17:53.22 | WIMPy | You don't know if it's going to be a transfer before it completes. |
| 17:53.28 | v0lZy | the receiver still sees only his number |
| 17:53.43 | WIMPy | Works in the current version. |
| 17:53.45 | v0lZy | i dont get that updated |
| 17:53.48 | WIMPy | NFI since when, however. |
| 17:53.55 | v0lZy | not on aastra and not on draytek |
| 17:54.22 | WIMPy | I'd try a new Asterisk :-) |
| 17:54.52 | v0lZy | but mine is Asterisk 1.8.4.4 built by root @ on a i486 running Linux on 2012-04-25 16:59:15 UTC |
| 17:54.54 | v0lZy | relatively new |
| 17:55.10 | leifmadsen | that's pretty old actually |
| 17:55.12 | WIMPy | Err, no. |
| 17:55.19 | v0lZy | what version is out now? |
| 17:55.35 | leifmadsen | asterisk.org will tell you |
| 17:55.48 | WIMPy | Or the channels topic. |
| 17:56.39 | *** part/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 17:57.05 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 17:57.14 | v0lZy | 10.6.1??? |
| 17:57.16 | *** join/#asterisk ctaloi (~ctaloi@50-56-202-179.static.cloud-ips.com) |
| 17:57.21 | v0lZy | from 1.8 to 10?!?!!?! |
| 17:57.25 | leifmadsen | that's a different major version |
| 17:57.30 | leifmadsen | 1.8.14.1 |
| 17:57.31 | WIMPy | That'd be worth a try. |
| 17:57.46 | leifmadsen | and at an avg of 4-6 weeks per release... 1.8.4 is old |
| 17:58.26 | WIMPy | If you start an attended transfer early and make it effectively a blind transfer, the display isn't updated, however. Only once the call is answered. |
| 17:58.36 | WIMPy | So still room for improvement. |
| 17:59.12 | WIMPy | Actually not that unimportant I think. |
| 17:59.14 | v0lZy | ha |
| 17:59.17 | v0lZy | not in my case |
| 17:59.25 | v0lZy | i can start a transfer |
| 17:59.33 | v0lZy | and whenever i decide not to wait for the party to hangup |
| 17:59.39 | v0lZy | the other side will get blind transfer |
| 17:59.45 | v0lZy | and its gonna transmit everything aok |
| 18:00.03 | WIMPy | At that point? |
| 18:00.18 | leifmadsen | all the points! |
| 18:00.51 | WIMPy | Yes, doing it via sip kind of works. |
| 18:01.19 | WIMPy | It will ternonate the call and set up a new call, but after some ugly delay. |
| 18:01.24 | v0lZy | WIMPy: yes |
| 18:01.47 | v0lZy | i put the user on hold and he gets the music while i dial the other extension |
| 18:01.56 | v0lZy | now as long as im ringing |
| 18:02.02 | *** join/#asterisk italorossi (~italoross@201.76.151.254) |
| 18:02.03 | v0lZy | the other extension sees my name and my number |
| 18:02.18 | v0lZy | as soon as I drop it, they see the original number and my number |
| 18:02.25 | v0lZy | then they pick up the phone. |
| 18:02.50 | WIMPy | Works for me as well, but there's a huge delay at that point. |
| 18:03.05 | v0lZy | on the other hand, before the editing today, if i held out until they picked up the phone and then dropped it, they only saw my name and number even after i dropped |
| 18:03.24 | WIMPy | And also a missed call. |
| 18:03.36 | v0lZy | erm no, im not explaining well |
| 18:04.08 | v0lZy | the desticntionbetween blind and attended transfer in my case is just if I wait for the user to pickup or not |
| 18:04.32 | v0lZy | if i drop the phone, before he answers, his phone keeps ringing with the original callers id |
| 18:04.41 | WIMPy | Yes, that's exactely what I'm talking about. |
| 18:04.48 | v0lZy | if i dont drop the phone and wait for him to pickup, he sees my ID until i drop the phone. |
| 18:04.56 | v0lZy | now without what i did today |
| 18:05.08 | v0lZy | after i would have dropped the phone, tehy would still see my id only, but talk to the original caller |
| 18:05.17 | WIMPy | For me the call is ended and after a few moment he gets another call with the original callers number. |
| 18:05.21 | v0lZy | it would not change to the original caller id when i got off the line. |
| 18:05.38 | v0lZy | uh.. no |
| 18:05.42 | v0lZy | in my case this is all instantanious |
| 18:05.56 | v0lZy | as soon as i drop my line while having the other person waiting with some music |
| 18:06.18 | v0lZy | their music cuts out for them, and for the person i did attended transfer to, i just get replaced by the original caller |
| 18:06.22 | v0lZy | no lag |
| 18:06.53 | WIMPy | has a very noticable lag. |
| 18:07.02 | WIMPy | And a missd call at the destination. |
| 18:08.34 | WIMPy | And no updated for the caller. But I could live with that. |
| 18:09.52 | v0lZy | lag |
| 18:10.02 | v0lZy | and a missed call at the destination you are transfering to?!!?! |
| 18:10.15 | WIMPy | yes |
| 18:11.10 | WIMPy | That's if it's done via SIP. If done via features that's not the case, but then the caller ID isn't updated ultil the call is answered. |
| 18:12.30 | v0lZy | i think youd benefit from what i have here |
| 18:12.40 | v0lZy | my current setup |
| 18:12.43 | v0lZy | passes 2 things |
| 18:12.54 | v0lZy | CallerID(name) and CallerID(number) |
| 18:13.07 | v0lZy | where name is the original caller |
| 18:13.13 | v0lZy | and number is the person transfering |
| 18:13.21 | v0lZy | now, if i do transfers |
| 18:13.56 | v0lZy | the user getting the transfer sees that im doing an attended transfer |
| 18:14.02 | ideaman | Anyone: In my older versions of Astersisk, for PRI, I always had chan_dahdi PRI signalling set to 'signalling= pri_net', since we're the provider, but now I get this error in svn compiled of 1.8 and dahdi 2.6.1....Thoughts? [Jul 25 12:08:57] ERROR[1490]: chan_dahdi.c:17356 process_dahdi: Unknown signalling method 'pri_net' at line 26. |
| 18:15.03 | WIMPy | Sounds like a hack. It's better already with onboard features. But indeed if the attended transfer becomes blind you lose the information about ther transferrer. |
| 18:15.06 | v0lZy | hm |
| 18:15.08 | v0lZy | it is a hack |
| 18:15.11 | v0lZy | and i just noticed something |
| 18:15.13 | v0lZy | now a user doesnt know |
| 18:15.22 | v0lZy | if the call is attended transfer or blind transfer |
| 18:16.06 | v0lZy | but at least they know its a transfer |
| 18:16.31 | v0lZy | and we usually do blind transfers anyway |
| 18:16.34 | WIMPy | For me the user doesn't know know the call is a transferred call if it was blind. But I think that may be a chan_sip issue. |
| 18:16.39 | WIMPy | checks |
| 18:18.07 | WIMPy | Nope. Not restricted to sip. |
| 18:21.33 | v0lZy | we seem to have opposite problems ::D |
| 18:21.40 | WIMPy | Yes |
| 18:24.30 | v0lZy | you get what i want |
| 18:24.44 | v0lZy | but lack some stuff i want to keep |
| 18:25.41 | WIMPy | Maybe some day... |
| 18:25.46 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-17.phlapa.fios.verizon.net) |
| 18:26.17 | v0lZy | maybe i should invert the two.... |
| 18:31.51 | WIMPy | still wonders if it's possible to send a correct caller id when using feature transfers. |
| 18:35.31 | karl370 | I've been working on setting up BLF on my asterisk system. I have hints working..it shows the proper state when I do a "core show hints". Watches shows 0. Now I need to subscribe fron another phone. I'm using Polycom 650's. To subscribe, does that mean simply register a specific line for the extension that I want to watch? |
| 18:36.39 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:39.54 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:48.48 | *** join/#asterisk eicto (~eicto@eicto.broker.freenet6.net) |
| 18:50.08 | WIMPy | learns that the whole topic has been heavily discussed 5 years ago. So maybe we get the other hlf of that by 2017. |
| 18:53.06 | drmessano | lol |
| 18:53.51 | WIMPy | does not find that funny at all. |
| 18:54.09 | drmessano | Not even a little? |
| 18:54.31 | WIMPy | Nope |
| 18:55.29 | WIMPy | But I find companies promising you that Asterisk can do everything your old PBX could do very interesting. |
| 18:56.11 | drmessano | You would think a simple patch would allow a T appended to transfers or something |
| 18:56.31 | WIMPy | Pardon? |
| 18:56.51 | Nugget | would gladly pay you Tuesday for a BLF today. |
| 18:56.55 | [TK]D-Fender | karl370, No, it means enabling presence on the Polycom (it isn't by default, this requires PROVISIONING them), then adding teh contacts, enabling "Buddy Watch" on them, and then making sure your hints are linked to the context the phone points to or "subscribecontect" |
| 18:57.17 | drmessano | Maybe I am on the wrong topic |
| 18:57.31 | drmessano | Thats what I get for letting myself get sucked into a meeting |
| 18:57.51 | drmessano | "Fine, I will be there.. but I am going to IRC from my iPad" |
| 18:59.34 | *** join/#asterisk eject_ck (~eject@213.159.242.42) |
| 19:00.56 | drmessano | I think Asterisk was built to support a subset of features to build the PBX of the future, and the last 5 years have been spent adding features of the past. Discuss |
| 19:01.17 | *** join/#asterisk eject_ck (~eject@213.159.242.42) |
| 19:02.09 | karl370 | [TK]D-Fender: Thanks. I have a central provisioning server. I've upgraded to the latest firmware/bootrom. It changed the format of all the config files. I thought that there was a more central way instead of adding a contacts on each phone & then doing the buddy watch thing. |
| 19:02.42 | [TK]D-Fender | karl370, There are some EFK ways of doing this, but it's kinda messy |
| 19:02.58 | [TK]D-Fender | karl370, if you're on a 3.3+ firmware on a suitable phone. |
| 19:02.59 | karl370 | EFK? |
| 19:03.05 | [TK]D-Fender | Extended Feature Key |
| 19:03.12 | [TK]D-Fender | not stuff |
| 19:03.14 | [TK]D-Fender | new* |
| 19:04.43 | WIMPy | drmessano: "some of the features everyone took for granted before" |
| 19:04.54 | karl370 | It seems to me that maybe that would be better than having do add contacts on each phone.?? I'm using asterisk 10. My phones are all Polycom 650's, with the lastest firmware/bootroms. |
| 19:06.05 | karl370 | I will look into EFK. |
| 19:06.16 | *** join/#asterisk twanny796 (~twanny@46.11.14.192) |
| 19:09.32 | [TK]D-Fender | better with contacts on the phone |
| 19:09.38 | [TK]D-Fender | far easier |
| 19:12.18 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 19:12.43 | karl370 | [TK]D-Fender: ok, I'll heed your advice. So I'll go down the contact route. I'll read up on the steps I need to do. Thank you. |
| 19:16.18 | Katty | HAI LADS |
| 19:17.41 | *** join/#asterisk eject_ck (~eject@213.159.242.42) |
| 19:17.47 | leifmadsen | Katty: o |
| 19:17.48 | leifmadsen | o/ |
| 19:20.36 | *** join/#asterisk eicto (~eicto@eicto.broker.freenet6.net) |
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| 19:23.39 | chuckf | gets some caffinated tea |
| 19:23.46 | *** join/#asterisk eject_ck (~eject@213.159.242.42) |
| 19:23.46 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 19:24.00 | ruben23 | hi guys how do i check status of zaptel on my asterisk server using sangoma cards |
| 19:26.42 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 19:27.37 | ectospasm | ruben23: zaptel hasn't been official for Asterisk systems in over four years. |
| 19:28.14 | ectospasm | with DAHDI you could do dahdi_scan, or cat /proc/dahdi/1 |
| 19:28.34 | ectospasm | ...but I can't remember for zaptel, unfortunately. |
| 19:29.40 | ectospasm | er, nearly four years |
| 19:29.47 | ectospasm | I exaggerated a smidge |
| 19:30.46 | Katty | steals drmessano's ball of computers |
| 19:30.53 | drmessano | lol |
| 19:31.10 | drmessano | 3 of the 4 survived the trip |
| 19:31.16 | Katty | woot! |
| 19:31.27 | Katty | i would not have been shocked if that MESS had no survivors |
| 19:31.31 | n3hxs | It's Katty! |
| 19:31.33 | drmessano | Nor would I.. |
| 19:32.25 | drmessano | One of them was hit so hard during shipping that the back door will not fit on because the opening is 1/2 inch smaller |
| 19:32.43 | *** join/#asterisk bchia (~Adium@nat/digium/x-vwvqacnahmnofizu) |
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| 19:33.48 | drmessano | This is why I argue so much when I get told to beg another market for their junk because we don't want to spend the money for a project |
| 19:33.52 | drmessano | Ta da! |
| 19:34.09 | drmessano | 3 out of 4.. Good thing I asked for extra! |
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| 20:04.43 | danfromuk | Hi, Anyone got experience provisioning Polycom phones over FTP? How do you protect the authentication details from being 'sniffed' thereby giving access to all the polycom config details to a potential hacker? |
| 20:05.24 | *** part/#asterisk pigpen (~mark@fw.seamans.cc) |
| 20:09.41 | leifmadsen | danfromuk: you don't -- that's the nature of FTP |
| 20:09.45 | *** join/#asterisk eject_ck (~eject@213.159.242.42) |
| 20:09.47 | leifmadsen | you use HTTPS or FTPS |
| 20:09.51 | leifmadsen | (SFTP) |
| 20:10.28 | danfromuk | Ok, i'll give HTTPS a bash. |
| 20:10.33 | danfromuk | Thanks for your reply. |
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| 21:00.11 | *** join/#asterisk Zopsi (1818afbd@gateway/web/freenode/ip.24.24.175.189) |
| 21:00.28 | Zopsi | Has anyone played around with digium phones? |
| 21:02.27 | WIMPy | You think they don't really exist? |
| 21:04.23 | Zopsi | Was just curious if you could change the ringtone through an alert-info for one call. |
| 21:04.44 | Qwell | Zopsi: not yet - but very very soon |
| 21:05.24 | WIMPy | Would be nice if you could change that in a channeltype independant way as well. |
| 21:05.46 | Zopsi | Hm.I thought they would have had that by now. |
| 21:06.00 | Qwell | Zopsi: when I say "very very soon", I mean really very seriously soon. |
| 21:06.25 | Qwell | malcolmd would probably revoke my release beers if I gave you an estimated date. :( |
| 21:06.42 | Zopsi | alright Qwell. I just put in a couple Algo SIP doorphones and it would be nice if they had a different ringtone. |
| 21:06.51 | malcolmd | yeah, you're at two beers on me, it'd cut you down to 1 or none |
| 21:06.56 | Qwell | heh |
| 21:07.29 | Zopsi | don't worry Qwell I'll acquire you more beer |
| 21:07.44 | Qwell | Zopsi: this is, of course, assuming you're configuring with DPMA (which you should be). It is currently possibly via the XML configuration. |
| 21:07.50 | malcolmd | correct |
| 21:08.23 | Zopsi | I'm using DPMA currently. |
| 21:08.50 | malcolmd | Qwell: it's been a while since i've been to the Nook; i told mike he could come too, but that he's on his own dime |
| 21:09.22 | malcolmd | Zopsi: just hold out a little longer, and it'll be easy to setup customized alerts for various ringing types w/ customized tones using the dpma |
| 21:10.08 | Zopsi | Alright I can probably hold out for another week or two, but then I have to switch to XML. |
| 21:16.26 | *** join/#asterisk eject_ck (~eject@213.159.242.42) |
| 21:24.41 | leifmadsen | luckily xml config isn't hard either :) |
| 21:26.49 | Qwell | No, but you get all the awesome features with DPMA. |
| 21:44.47 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 21:48.14 | Curs0r | When my extensions ring and the cid simply says 'asterisk' that's a call with bad cid info yes? |
| 21:48.45 | WIMPy | Probably none |
| 21:49.40 | Curs0r | The guppies are still complaining at me about that one, hehe. I don't know what to tell them. My calls seem to get routed fine with it. |
| 21:50.10 | ChannelZ | Maybe the server wants to talk to them. |
| 21:50.18 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 21:50.44 | Curs0r | These are also the users that just told me "it comes and goes" but were unable to cite any example of it "going" today |
| 21:52.26 | ChannelZ | I would say to them "my interest in your problem comes and goes as well," but I am of course mean. |
| 21:52.54 | leifmadsen | Curs0r: SIP? |
| 21:53.15 | Curs0r | I am getting mean I find. Ultimately they're paying me more to fiddle with this pstn card than they would just converting to sip trunking. |
| 21:53.35 | Curs0r | leifmadsen, sip extensions, tdm410 pstn card |
| 21:53.48 | leifmadsen | Curs0r: in sip.conf try sendrpid=yes |
| 21:54.07 | Curs0r | Doing that right now |
| 21:54.13 | ChannelZ | I use a little script to change blank CID to "Unknown" and do some local CID lookups |
| 21:54.16 | Curs0r | Just under general? |
| 21:54.24 | leifmadsen | ya |
| 21:54.53 | Curs0r | Done |
| 21:55.02 | *** join/#asterisk uskerine (~Uske@101.Red-2-138-191.dynamicIP.rima-tde.net) |
| 21:55.05 | uskerine | hi |
| 21:55.33 | uskerine | could anyone help me to look where i should start while installing digium TE121 card (for ISDN PRI) |
| 21:55.36 | uskerine | ? |
| 21:56.11 | jpsharp | Install the card, install DAHDI, install libpri, install asterisk. Config all. |
| 21:57.08 | uskerine | ok |
| 21:57.11 | uskerine | i have installed the card |
| 21:57.14 | uskerine | i have installed DAHDI |
| 21:57.17 | uskerine | i am not sure about libpri |
| 21:57.22 | uskerine | how can i check? |
| 21:57.23 | uskerine | :) |
| 21:57.25 | vastina | did you do a dundi test |
| 21:57.31 | vastina | sorry, bad joke |
| 21:57.37 | uskerine | yes i did |
| 21:57.41 | uskerine | and card is recognized |
| 21:57.46 | uskerine | but it seems it is configured for T1 |
| 21:58.20 | uskerine | i want to preconfigure it for E1, and any advice on what should I expect in terms of troubleshooting once provider provisions the E1 |
| 22:00.06 | vastina | not really, 30 bearer channels with a delta, much the same as 23 bearers with 1 delta... just set up the signalling to your intended country |
| 22:00.23 | uskerine | ok |
| 22:00.53 | uskerine | where should i start vastina? i have configured asterisk with a 4 port analog card but i am by far not an expert in asterisk |
| 22:01.17 | WIMPy | Data, not delta. |
| 22:02.11 | vastina | WIMPy: neither am I, http://www.techopedia.com/definition/26018/delta-channel-d-channel |
| 22:02.21 | vastina | s/neither am I,// |
| 22:02.22 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 22:02.26 | ideaman | Anyone: In my older versions of Astersisk, for PRI, I always had chan_dahdi PRI signalling set to 'signalling= pri_net', since we're the provider, but now I get this error in svn compiled of 1.8 and dahdi 2.6.1....Thoughts? [Jul 25 12:08:57] ERROR[1490]: chan_dahdi.c:17356 process_dahdi: Unknown signalling method 'pri_net' at line 26. |
| 22:02.43 | vastina | uskerine: that was for you... neither am I an expeter, learning a lot |
| 22:02.54 | vastina | WIMPy's been somewhat helpful to me :) |
| 22:03.36 | vastina | s/expeter/expert/ |
| 22:03.42 | vastina | wow I can not type |
| 22:03.48 | jpsharp | ideaman: Did you build libpri along with 1.8 and DAHDI? |
| 22:04.11 | ideaman | ahh, didn't build, just apt-get |
| 22:04.12 | ideaman | dangit |
| 22:04.16 | ideaman | I knew that too |
| 22:04.23 | jpsharp | And specifically, did you build libpri first. |
| 22:04.38 | ideaman | I built dahdi, then installed libpri, then built asterisk |
| 22:04.50 | ideaman | as long as I was building all, is that the right order still? |
| 22:05.26 | jpsharp | I don't remember if there are dependencies in dahdi versus libpri, but I've always build libpri first, then dahdi, then asterisk. |
| 22:05.34 | ideaman | k |
| 22:05.40 | ideaman | thanks, I'll go try that |
| 22:07.41 | WIMPy | vastina: Unless you can come up with an ITU or ETSI document thent mentions the word "delta" anywhere I clain that link to be wrong. |
| 22:08.46 | WIMPy | As far as I know libpri is only used by chan_dahdi, I/E/ Asterisk. |
| 22:10.23 | jpsharp | I couldn't remember if you *had* to build libpri first, but now that I remember some of the dahdi source code, there's #ifdefs for libpri, so you hhave to build libpri first. |
| 22:11.16 | jpsharp | Funny that I can remember chunks of source code from years ago, but can't remember what I went to the grocery store for. |
| 22:11.30 | WIMPy | :-) |
| 22:13.01 | Curs0r | jpsharp, that happens. I can quote you 23 years of simpsons episodes verbatim, but I always have to verify the t568b pinout when making ethernet cables |
| 22:13.46 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 22:13.51 | Nivex | w/orange orange w/grn blu w/blu grn w/brn brn |
| 22:13.55 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-dloiohbyiplouoss) |
| 22:13.57 | Nivex | yes that was from memory |
| 22:14.03 | Curs0r | :) |
| 22:15.55 | uskerine | there is a jumper in the card |
| 22:16.04 | uskerine | to select T1 or E1 |
| 22:16.34 | WIMPy | In later dahdi versions you can do it in software as well. |
| 22:17.09 | uskerine | i am using binary repository from ubuntu server 12.04 |
| 22:19.01 | jpsharp | Are the binary packages built with ISDN support? |
| 22:19.46 | jpsharp | Knows not any thing about packages. |
| 22:20.24 | vastina | WIMPy: meh https://en.wikipedia.org/wiki/D_channel why would wiki call it a delta channel? and I spent way too much time looking this up but apparently it was originally called a delta channel as it was a "delta" to the bearer channels, under all ITU documentation I can find I can only find D, so neither data or delta are official under the Q.931 |
| 22:20.29 | vastina | anyway |
| 22:21.20 | vastina | i still have no idea why my vtech phone is still appearing as unreachable under CLI when I can send receive calls from it, but I can't send an MWI indication to it |
| 22:21.29 | vastina | less trivial and much more important to me |
| 22:22.03 | vastina | i did change change the qualifyfreq |
| 22:22.07 | vastina | no change |
| 22:22.10 | Curs0r | The vtech doesn't do its own mw polling if you assign it a mailbox in sip.conf |
| 22:22.11 | Curs0r | ? |
| 22:22.48 | vastina | Curs0r: I don't think so, looking all over the configuration front-end |
| 22:23.20 | Curs0r | I got lucky with that, the spa942 took its cues from that line and I didn't have to do anything else |
| 22:23.50 | vastina | mine is a S1210 |
| 22:23.56 | vastina | a hotel phone |
| 22:24.10 | WIMPy | vastina: I still think that once someone made a mistake and a lot of others just copied it. |
| 22:24.34 | vastina | WIMPy: talking about the data/delta argument? |
| 22:24.46 | WIMPy | yes |
| 22:25.06 | vastina | to me it's about as frutiful as a vi/emacs debate... all good so long as we know what eachother is speaking about i suppose |
| 22:25.29 | uskerine | nerver heard about anything but "D" channel |
| 22:25.31 | WIMPy | yes |
| 22:25.41 | WIMPy | But delta implies a difference but to what? |
| 22:25.47 | vastina | to bearer |
| 22:25.50 | *** join/#asterisk pyther (~pyther@unaffiliated/pyther) |
| 22:25.53 | vastina | delta to bearer or B channel |
| 22:26.18 | WIMPy | That just doesn't make sense. |
| 22:26.46 | vastina | it's what I read, and frankly I could give a rat's * :p |
| 22:27.40 | vastina | call it a dicking channel for all I care, but I do want this vtech working! |
| 22:28.12 | WIMPy | Buy Q.931 for penis enlargement. |
| 22:28.34 | vastina | :) |
| 22:29.06 | Curs0r | But what about my enzyte pen? I never go anywhere without my pressurized placebo injector |
| 22:29.22 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a09e9.pool.mediaWays.net) |
| 22:29.58 | WIMPy | Blue book instead of blue pill. |
| 22:30.08 | WIMPy | (was it at that time?) |
| 22:36.51 | *** join/#asterisk egonzalez_ergio (~egonzalez@186.137.1.3) |
| 22:40.00 | egonzalez_ergio | Hi: I'm using AsteriskNOW. I've created a stream in Music On hold , but dont know how to force this strean to be the defalut MoH for all extensions |
| 22:40.39 | Curs0r | Naming it [default] worked for me |
| 22:40.55 | egonzalez_ergio | In Asterisk SIP settings I've added musiconhold=streamName |
| 22:40.59 | egonzalez_ergio | but don't work |
| 22:41.42 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-18-80.bois.qwest.net) |
| 22:42.10 | egonzalez_ergio | Hi Curs0r : I can't rename fron FreePBX gui and, if I rename the strean in the config file, after reboot the file is rewrited with the original settings |
| 22:45.08 | Curs0r | Aaand now this pbx isn't accepting incoming calls again... marvelous |
| 22:45.36 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
| 22:46.59 | *** join/#asterisk ChrisInSydneyToo (~Chris@60-242-81-231.tpgi.com.au) |
| 22:48.36 | Curs0r | Direct dialing one of the other numbers in the hunt_did_ work... this sick, confusing, freakish SOB |
| 22:49.21 | Curs0r | It's true what they say. Life is much harder for us stupid folk |
| 22:50.26 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 22:51.14 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
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| 23:22.51 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-jqwgdhuwqoshjbgp) |
| 23:25.00 | *** join/#asterisk danfromuk (~IceChat77@2.27.12.118) |
| 23:25.27 | danfromuk | Hi, Has anyone got https provisioning working with Polycom IP3XX phones? |
| 23:27.03 | *** part/#asterisk egonzalez_ergio (~egonzalez@186.137.1.3) |
| 23:40.55 | leifmadsen | yes |
| 23:41.08 | leifmadsen | doesn't require a whole lot... other than for the https server to provide data to the phone |
| 23:47.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 23:54.55 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |