| 00:26.20 | *** join/#asterisk bcrisp (~bcrisp@70.102.242.138) |
| 00:26.24 | bcrisp | hi all |
| 00:27.55 | bcrisp | I'm having an issue with outbound softphone calls, calls drop in ~ 20-30 seconds, think its a NAT issue (soft phones are behind nat firewall), * server is not. Can anyone provide a link to some troubleshooting resource on this? |
| 00:28.17 | carrar | ~sipnat |
| 00:28.18 | infobot | somebody said sipnat was Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:28.24 | bcrisp | thanks |
| 00:33.46 | *** join/#asterisk coppice (~chatzilla@m121-203-197-103.smartone-vodafone.com) |
| 00:36.10 | *** join/#asterisk dwayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 00:43.42 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 00:49.57 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 00:55.28 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 00:57.12 | *** join/#asterisk russ (~russ@206.29.188.239) |
| 01:13.14 | *** join/#asterisk mweichert (~mweichert@CPE687f749ffa09-CM0011ae8a737a.cpe.net.cable.rogers.com) |
| 01:15.08 | mweichert | Hi, I'm trying to use the Goto application with a variable in the context - Goto(ivr-${IVR_ID},s,1) but that doesn't work - the variable is never interpolated. When I include "ivr-${IVR_ID}", the variable get's interpolated but "ivr-4" (with quotes) is not a valid context. Any suggestions? |
| 01:16.25 | *** join/#asterisk russ (~russ@206.29.188.191) |
| 01:22.09 | *** join/#asterisk b0gatyr (~b0gatyr@adsl-10-92-8.mia.bellsouth.net) |
| 01:27.25 | upb | interpolated ?:PPP |
| 01:27.52 | bcrisp | this is getting on my nerves |
| 01:29.44 | carrar | Kick it |
| 01:30.01 | bcrisp | 30 seconds on the dot every time |
| 01:30.11 | carrar | hit it with a hammer |
| 01:30.14 | bcrisp | i want to |
| 01:30.20 | carrar | do it!! |
| 01:30.26 | carrar | everyone is doing it!! |
| 01:30.29 | carrar | peer pressure!! |
| 01:30.31 | bcrisp | im not sure how to troubleshoot it |
| 01:31.10 | upb | lsniff the network to see what causes the drop ? |
| 01:33.49 | bcrisp | oh crap i think its the softphone setting |
| 01:34.12 | bcrisp | phone seems to think rtcp is inactive possibly |
| 01:35.29 | upb | better verify what is actually happening and which party is breaking the connection etc |
| 01:36.27 | bcrisp | it seems to resolve thie issue - xlite advanced options, "In times of network disruption, automatically hang up calls after: (RTCP has been inactive for 30 seconds" |
| 01:36.32 | bcrisp | the hangup after 30 seconds was the default |
| 01:39.48 | bcrisp | yep it fixed it |
| 01:41.24 | bcrisp | so its confused |
| 01:41.50 | bcrisp | thamks |
| 01:41.52 | bcrisp | thanks |
| 01:50.23 | mweichert | when using "goto" to jump to a different context, what happens to the channel variables? |
| 02:01.36 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 02:08.20 | *** join/#asterisk GoRK (~gork@c75-111-94-145.amrlcmta01.tx.dh.suddenlink.net) |
| 02:09.06 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 02:13.17 | GoRK | is there a version of digium's g729 codec that works with 1.8 yet? |
| 02:23.28 | *** join/#asterisk grolloj (~chatzilla@cpe-98-14-29-246.nyc.res.rr.com) |
| 02:32.37 | ChannelZ | don't think so |
| 02:34.21 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 02:37.48 | GoRK | thats pretty ridiculous if so |
| 02:43.01 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 02:43.33 | pabelanger | GoRK: No, they will release a version once 1.8.0 is released |
| 02:46.09 | coppice | It is ridiculous. It reduces the testing 1.8.0 gets before release, as many people need G.729 to be able to set it up and try it |
| 02:46.43 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 02:48.57 | pabelanger | Ridiculous? How? 1.8.0 is still in beta, API / ABI are still subject to change. g729 codec is a product, takes time and energy to build / maintain. |
| 02:49.09 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:49.45 | coppice | if the API hasn't stabilised the beta is a ridiculous name for it |
| 02:49.52 | *** join/#asterisk sbrath (~sbrath@unaffiliated/sbrath) |
| 02:51.01 | carrar | ridiculous!! |
| 02:51.02 | sbrath | I know fax is s sore subject, but what components do I need to make an email->fax->T38 Gateway solution. I don't have any hard lines, but I have a T.38 compatible provider.. Will AsterFax deliver via T.38 ? |
| 02:51.27 | pabelanger | carrar: Ridiculous!!1!one! |
| 02:51.36 | carrar | !!!1!!I!!i!! |
| 02:51.46 | carrar | !l!!! |
| 02:56.55 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-154-222.dsl.stlsmo.sbcglobal.net) |
| 02:57.14 | LemensTS | whats the best way to prevent against sip phone password attacks |
| 02:57.44 | bougyman | don't use sip |
| 02:57.46 | sbrath | don't put your asterisk box directly on the internet. |
| 02:57.59 | bougyman | two good ways, there. |
| 02:58.14 | sbrath | use a Session Border Controller... |
| 02:58.23 | coppice | the death penalty for password attacks |
| 02:58.24 | florz | erm ... |
| 02:58.28 | sbrath | Pick really good passwords. |
| 02:58.36 | florz | LemensTS: use proper passwords |
| 02:58.49 | florz | it's simple, isn't it? |
| 02:58.52 | sbrath | monitor logs for password failures, and block IPs with iptables. |
| 02:58.58 | florz | sbrath: NO |
| 02:59.16 | florz | we don't really have to discuss this every week, do we? |
| 02:59.23 | sbrath | which thing? |
| 02:59.35 | sbrath | passwords or fax? |
| 02:59.38 | bougyman | we use openvpn for any external access other than trunks that use IP auth from providers for inbounds. |
| 02:59.48 | florz | the idea of adding a DoS vulnerability in order to prevent a non-vulnerability |
| 03:00.01 | bougyman | doesn't stop someone internally from trying to sniff or brute force passwords, though. |
| 03:00.14 | LemensTS | yea im getting thousands of attacks bring things to a creep... |
| 03:00.38 | LemensTS | Not that concerned with someone getting password |
| 03:00.42 | bougyman | we never put a 5060 public on the internet |
| 03:00.55 | bougyman | changing that port, even for inbound provider traffic, is a good idea. |
| 03:01.10 | florz | LemensTS: well, ok, in that case verify manually that the ip address is not a legitimate user and block it |
| 03:01.36 | LemensTS | cant change port got too many voip adapters out there that arent provisioned |
| 03:01.49 | LemensTS | florz: its from germany definately a bot |
| 03:02.01 | bougyman | LemensTS: all of them have an option to use a different port. |
| 03:02.08 | sbrath | florz: which thing did I mention that exposes a DoS attack surface, any different than a asterisk directly on the net? |
| 03:02.12 | florz | LemensTS: I can tell you there are not just bots in .de =:-) |
| 03:02.12 | bougyman | you ahve to provision _something_, the port is just one more thing. |
| 03:02.20 | LemensTS | ive read about scripts on counting the number of attempts then blocking the ip in iptables....think that is what u are talking about also |
| 03:02.28 | bougyman | port knocking? |
| 03:02.44 | bougyman | none of your devices would support that without a fw in front of them that understood port knocking. |
| 03:02.52 | florz | sbrath: well, maybe I did just interpret what people usually suggest: you did mean to automate this monitoring and blocking, didn't you? |
| 03:02.54 | bougyman | by then you might as well have openvpned in. |
| 03:03.20 | sbrath | yes, watching logs, and scripting to have iptables blackhole anoyances. |
| 03:03.39 | florz | sbrath: that is obviously a DoS vulnerability |
| 03:04.22 | carrar | adapt a open root policy |
| 03:04.41 | sbrath | I guess if someone fakes the source IP against you and then you block it, sure. I've really never had the block be automatic, just log, and manually add later. |
| 03:05.26 | sbrath | As well as configure the software to exclude source ip's in known trusted blocks, then the worst I'd do is block someone from home I guess. |
| 03:06.04 | sbrath | So now that the channel is buzzing, any words of advice to configure a email -> asterisk -> T38 provider -> Fax ? |
| 03:06.27 | sbrath | t38modem + asterfax ? Will that do it? |
| 03:06.40 | carrar | Just block all none USA IP's :) |
| 03:06.57 | sbrath | does someone maintain a list of non-us IP's ? |
| 03:06.59 | carrar | about 80 lines in iptables |
| 03:07.00 | coppice | asterfax is now noojeefax (dunno about that spelling) and I would avoid it |
| 03:07.03 | carrar | yes |
| 03:07.08 | carrar | see ARIN |
| 03:07.17 | florz | sbrath: well, yeah, that's fine, of course - but people often suggest blocking things automatically, and completely forget about the fact that (a) the attacker is not authenticated, therefore you can not block the attacker and (b) it's pointless to defend this way against passwords getting cracked |
| 03:08.20 | carrar | sbrath: http://www.iana.org/assignments/ipv4-address-space/ipv4-address-space.xml |
| 03:08.43 | sbrath | how nice of IANA |
| 03:09.01 | sbrath | What about HylaFax ? |
| 03:10.13 | coppice | there are a number of email to fax add ons for *. use one of those with the built in T.38 support |
| 03:14.09 | *** join/#asterisk Kyosh (~whoa@96.246.232.130) |
| 03:14.28 | sbrath | any youve tried? All look so imature.. |
| 03:18.13 | LemensTS | i check every minute to see if sip channels is above 100, if so then it text's me....this seems easiest for catching problems in asterisk from what i have found |
| 03:19.11 | drmessano | I just check to see what time it is in China.. if it's between 00:00 and 23:59, I am probably being attacked |
| 03:19.25 | LemensTS | lol |
| 03:20.11 | mweichert | hello... using AMI, I'm not receiving hangup events... most other events seem to be displayed. I'm using EVENTFLAGS="on" ... any ideas? do I have to enable them explicitly somehow? |
| 03:28.41 | coppice | drmessano: its 11:28AM in China, and we are being flooded with spam from US mail servers |
| 03:33.28 | drmessano | Servers made in China |
| 03:33.43 | drmessano | It's a vicious cycle |
| 03:33.43 | coppice | are servers made anywhere else? |
| 03:34.13 | coppice | actually, quite a lot of their power supplies are made in Thailand |
| 03:40.03 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 03:49.16 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 03:52.30 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 03:56.45 | *** join/#asterisk [netman] (~netman@198.Red-81-47-144.staticIP.rima-tde.net) |
| 03:58.17 | russ | If I forward unanswered calls from my cell phone to my teliax DID, is there any possible to tell if an incoming call is a forward from my cell, or a call directly to the DID? |
| 04:01.16 | jly2680 | someone here had asstrra phones work with their asterisk box? |
| 04:01.17 | carrar | probably not |
| 04:01.22 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:01.31 | carrar | unless you see a diversion header, which I doubt |
| 04:02.00 | carrar | but it wouldbe the cell phone creating that so your carrier probably hides that |
| 04:03.24 | russ | I have a packet capture, let me have a look |
| 04:03.43 | *** join/#asterisk adolfomaltez (~taro@190.87.99.130) |
| 04:03.54 | jly2680 | if someone here can give me a aastra 4422 bootrom and application file |
| 04:04.21 | russ | am I just looking in the IAX "NEW" packet? |
| 04:06.35 | carrar | jly2680, why not download it yourself? http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-9D124659/03/hs.xsl/21669.htm |
| 04:07.10 | carrar | 4422 is a release? |
| 04:07.19 | jly2680 | 4422 is not included on aastra site |
| 04:08.05 | jly2680 | will i need this file to boot 4422?i need to set up a web server? |
| 04:08.09 | russ | What is "call identifier"? |
| 04:08.26 | russ | does that just index the call within the trunk? |
| 04:19.25 | jly2680 | how to solve erroe unreachable client |
| 04:22.39 | *** join/#asterisk corretico (~laguilar@201.201.44.82) |
| 04:24.08 | kerframil | exit |
| 04:24.10 | kerframil | exit |
| 04:24.44 | [TK]D-Fender | jly2680: Have if become contactable. Or register |
| 04:25.32 | jly2680 | every 3 minutes my client becomes unreachable |
| 04:26.33 | jly2680 | my asterisk is behind a nat server and my client too |
| 04:26.47 | [TK]D-Fender | jly2680: Get a more stable connection or client, or increase your qualify time |
| 04:27.48 | jly2680 | what is the default time for qualify time? |
| 04:28.08 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 04:28.16 | [TK]D-Fender | jly2680: 2000 ms |
| 04:28.31 | jly2680 | how can i increase it? |
| 04:28.34 | titter | jly2680: what type of client? |
| 04:29.05 | jly2680 | sip client..bm622 with voip ata |
| 04:31.03 | titter | jly2680: I have had some issues with clients dropping behind certain firewalls with my Polycom phones, however lowering the NAT keep alive has fixed the stability and the phones will stay registered with much more stability |
| 04:31.18 | [TK]D-Fender | QUALIFY=# |
| 04:31.25 | titter | jly2680: This setting was a specific setting to the Polycom firmware, not * |
| 04:31.34 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 04:32.18 | jly2680 | so i must adjust it on my client side? |
| 04:32.29 | [TK]D-Fender | jly2680: SIP PEER |
| 04:34.16 | jly2680 | qualify=3000ms? |
| 04:34.21 | *** join/#asterisk bdfoster (523b0c57@gateway/web/freenode/ip.82.59.12.87) |
| 04:34.23 | [TK]D-Fender | sure |
| 04:34.50 | jly2680 | il try now |
| 04:34.53 | jly2680 | thanks |
| 04:35.14 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 04:35.30 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-154-222.dsl.stlsmo.sbcglobal.net) |
| 04:37.50 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 04:37.58 | *** join/#asterisk nightwalk (~nightwalk@daimon.vixel.org) |
| 04:38.13 | Micc | I lower the registration expire down to 120 seconds to keep phones registered behind firewalls. |
| 04:38.15 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 04:38.29 | Micc | It could be higher, but 120 seems to be good enough for all routers I've encountered so far. |
| 04:38.58 | Micc | its also good if there is a server failure, they'll know sooner and try their backup. |
| 04:39.38 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:41.14 | titter | Micc: My major problem was with what the Sonicwalls do when SIP transformations was set to on. It mangles the SDP. Unfortunately some of these shared office spaces have really strict rules, or terrible I.T. companies which makes supporting SIP phones behind their NAT a real pain. |
| 04:42.04 | [TK]D-Fender | titter: Which is the very first thing we tell you to disable. |
| 04:42.48 | titter | [TK]D-Fender: Don't have to tell me ... try to explain that to an I.T. company that has no understanding of SIP but won't change the settings. |
| 04:44.17 | Micc | titter, we usually tell our customers to get new IT people if they aren't willing to make a simple change like that. |
| 04:44.43 | Micc | Then we recommend someone that knows how to do what we want. |
| 04:45.15 | titter | [TK]D-Fender: My company has decided that short term leases at shared office spaces is the way to go for all our regional offices. So I am now dealing with 23 different network and firewall configurations that are completely out of my control. Luckily I have only had a few issues, and the first office lease was the Sonicwall diaster. So every new lease goes thrugh a pretty much shake down. |
| 04:46.27 | Micc | titter, thats the key, to know what you need to know. In the begging its hard to know what you need to know until you run into it. |
| 04:46.54 | Micc | It does get easier. |
| 04:46.59 | titter | Ya, it was new to me that's for sure. I never ran into any issues with remote users and their equipment |
| 04:47.27 | titter | It is this entry level consumer stuff that seems to cause problems, simply due to the fact is isn't configured correctly 90% of the time |
| 04:48.13 | *** part/#asterisk adolfomaltez (~taro@190.87.99.130) |
| 04:48.48 | Micc | titter, in a lot of cases we replace the router/firewall when we install phones. |
| 04:49.09 | Micc | We put in our own specially configured router with bandwidth management. |
| 04:49.20 | Micc | QoS isn't enough in most cases. |
| 04:50.12 | titter | Micc: Most of these places won't allow us to do that, or want to charge us way more than what it is worth ... it would of been cheaper to rent our own building. Had one the other day want an extra $300/mo to do that |
| 04:51.15 | titter | Luckily these are small offices, so it at most is usually 5-6 phones |
| 04:53.21 | Micc | oh, your sharing an office with already established companies that don't want to change their phones and network? |
| 04:53.40 | titter | Bingo |
| 04:53.55 | Micc | ok then, you can't really ask someone to do something when there is no benefit to them. |
| 04:53.58 | titter | There are companies that buy buildings, and rent out the offices |
| 04:53.59 | Micc | Your in a bit of a pickle. |
| 04:54.00 | [TK]D-Fender | checkout time, later all |
| 04:54.17 | titter | So they sell us an office with x amount of data drops |
| 04:54.21 | titter | rent* |
| 04:54.26 | Micc | right. |
| 04:54.46 | titter | Ya, so it has been learning how to make the phones work beind their networks |
| 04:55.13 | Micc | thats a bummer if you can't make any changes to their router. |
| 04:55.29 | titter | Some networks are setup correctly, others not so much ... I had to explain VLAN's to one I.T. company and why their setup at an office was a violation of HIPPA |
| 04:55.29 | Micc | some phones might do better than others. |
| 04:55.53 | titter | The Polycom's seem to fair pretty well |
| 04:56.38 | titter | I have a cental provisioning server that I configure the phones to ... they download the specific config files per each phone. I made a web GUI to create these configs, and soon will integrate that directly witht he dialplan I have setup in * |
| 04:57.13 | titter | I basically have 6 servers around the country with 1 or 2 PRI's at each location |
| 04:57.34 | titter | I am using includes and building office specific dialplan conf files to keep things organized |
| 04:58.34 | titter | i.e. chicago.conf would include the extensinons for that office, the ivr, and a context for the sip to use for outbound calling rules ... soem places don't have to dial the full 10 digits for local calls, so I simulate that for them |
| 04:59.12 | titter | I also control outbound caller ID numbers this way, as well as E911 |
| 04:59.31 | titter | Seems to be working well, but I am sure I will find another way to do it in a few months and change it all |
| 05:01.32 | Micc | yeah, I use a ton of includes. |
| 05:01.58 | Micc | I try to keep all the ivr stuff seperate from outbound dial stuff and extensions and macros. |
| 05:02.46 | Micc | it wouldn't hurt to be in one big file though as long as you seperate it out for each location that probably keeps it organized enough. |
| 05:03.44 | Micc | what did you write the web gui in? |
| 05:04.02 | titter | asp.net |
| 05:04.28 | Micc | and you use tftp or http for provisioning? |
| 05:04.34 | titter | https |
| 05:05.10 | titter | If for some reason it is an older Polycom say a 501 I might have to use TFTP to get it to a bootrom that understands the newer syntax |
| 05:05.23 | titter | Once it updates I switch it back to https and ship the phone |
| 05:05.26 | Micc | I've been meaning to get around to making a phone provisioning web app for a while. I have scripts now. |
| 05:05.45 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 05:05.54 | titter | asp.net was just in front of me at the time, so I knocked it out quikcly ... it builds XML with a few simple lines of code |
| 05:06.17 | Micc | yeah, I'm a .net developer myself. |
| 05:06.22 | titter | Polycom is releasing their newest firmware soon, and the cfg files have changed ... again |
| 05:06.29 | Micc | working on a project with windows azure at the moment. |
| 05:06.33 | titter | Nice |
| 05:07.03 | titter | I like .net more than anything else I have messed with. I am a sysadmin by job title, but my hands are in everything including the PBX and all network admin |
| 05:07.06 | Micc | its been a bit of a learning curve, and some things don't work right like asmx on azure. |
| 05:07.18 | titter | Hmm |
| 05:07.25 | titter | That would be a curve lol |
| 05:07.47 | Micc | I'm exactly the opposite, I'm a programmer by job title, but I have my hands in sysadmin. |
| 05:08.27 | titter | I am actually trying to think if storing dialplan in a SQL db or Oracle db is a good idea |
| 05:08.30 | Micc | its not bad though once you finally get it working. I had a hell of a time just getting simple stuff to work. |
| 05:09.09 | Micc | you can use realtime with freetds and mssql server. |
| 05:09.13 | Micc | I did that for years. |
| 05:09.23 | titter | Thats like anything lol, learning the Polycom cfg files was a fun task, but compared to the Linksys phones I have messed with, the Polycoms are so much easier |
| 05:09.27 | Micc | but I never put the whole dialplan in there, just a few things. |
| 05:09.41 | titter | Ya I would just put these remote offices includes in there |
| 05:10.00 | Micc | yeah, I still don't care for polycom, but I haven't tried anything but the web config on linksys phones and atas. |
| 05:10.49 | titter | My boss is the CIO and he is an Oracle freak, so I would probably want to find a way to do it with Oracle |
| 05:11.18 | Micc | I don't remember how realtime works with extensions.conf but if it only hits the db on dialplan load, then it shouldn't be a problem. |
| 05:11.43 | Micc | I've only played with oracle on my home machine. |
| 05:12.22 | Micc | I'm sure theres an oracle ODBC client out there. |
| 05:12.38 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 05:12.49 | titter | Ya I believe it does just that, I never really got into the DB and * stuff yet. I remember talking about it at Digium, but for now it was just easier keeping it simple as I was taking over for our old PBX guy who let the * server get hacked and rack up over 300k of long distance charges in a weekend. He left a SIP account with the username and password as Administrator |
| 05:13.33 | titter | It was a 4 day weekend, and didn't have any type of notification sent to him |
| 05:13.52 | Micc | wow |
| 05:13.54 | titter | So they fired him, and sent me to Digium to crash course in the advanced class lol |
| 05:14.15 | titter | So I have been doing this for a little less than a year |
| 05:14.36 | titter | Yup all the calls were to Cuba |
| 05:14.59 | Micc | not bad, it took me 4 years to get really comfortable with asterisk, and I'm on 6 or 7 now and I still feel stupid sometimes. |
| 05:15.35 | titter | That class helped a lot .. looking at it now though, I think there should be another course for a truly advanced class |
| 05:15.39 | Micc | I never took a digium class though I would like to. |
| 05:16.13 | titter | Honestly the advanced class was more or less a very in depth introduction to Asterisk. The first day or so covered what was covered in the basic class |
| 05:16.20 | Micc | you can do some really cool stuff with func_odbc. |
| 05:16.49 | titter | I would like to find a stats program to monitor all calls on the system |
| 05:16.52 | Micc | I guess they expect people to jump right into advanced without doing the other classes. |
| 05:17.08 | Micc | we use one, but its not very good. |
| 05:17.43 | Micc | its good except when you want to know how many concurrent calls. |
| 05:18.12 | Micc | its called asterisk-stat |
| 05:18.44 | Micc | the version I have was written in 2005, so there has to be something better by now. |
| 05:18.47 | titter | Ya, I didn't take my dcap because it was more or less my first week with Asterisk ... but I feel I would of passed it. I did the pre-exam and had my practical knocked out with 30 minutes to spare. I had the dialplan almost written by the time asterisk was done compiling |
| 05:19.29 | titter | I was worried about the written side as that gets into a lot of voip and telephony jibberih like codecs |
| 05:19.51 | titter | I will check out that program |
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| 05:48.17 | ChannelZ | Anyone get this a fair amount: "RTCP SR transmission error to x.x.x.x:14829, rtcp halted Operation not permitted" |
| 05:49.40 | Micc | yup |
| 05:49.46 | Micc | probably about once a day |
| 05:50.34 | Micc | I've learned to ignore. Its just stats anyways. |
| 05:50.55 | Micc | And the IP addresses are usually one of my providers which I don't expect to support rtcp properly anyways. |
| 05:51.23 | ChannelZ | yeah I'm kind of puzzling over it. It will happen in the middle of a call, so I'm not sure what the 'Operation not permitted' is about (like it's not allowed to bind to the socket, but clearly it already is and is sending plenty of audio data fine). It's an ERROR class so I dunno about "just stats" |
| 05:52.00 | Micc | It doesn't seem to affect the call. |
| 05:52.36 | Micc | I think from what I remember reading the RTP traffic will work fine without it even. |
| 05:52.52 | titter | I have a weird issue ... I have a SIP trunk setup to a Shoretel system. When I dial one of the hunt groups on the Shoretel system from Asterisk, it places Asterisk into MOH, then goes crazy once the call is answered and no audio is transmitted |
| 05:54.16 | Micc | titter, no idea, I really don't care for shoretel, but I know that doesn't help you when you need to interface with it. |
| 05:54.59 | Micc | titter, have you looked at sip debug to see if it looks right? |
| 05:55.28 | titter | Ya it is weird ... the dialplan is 1 line ... a simple Dial to the SIP/shoresip/${EXTEN} |
| 05:56.03 | Micc | where is the MOH coming from? asaterisk or the shoretel? |
| 05:56.13 | titter | Asterisk |
| 05:56.17 | titter | That is what confuses me the most |
| 05:56.32 | titter | It is dialing Shoretel, Shoretel answers right away, but then MOH starts from Asterisk |
| 05:56.40 | Micc | have you tried forcing a ring in the dial command with the r option? |
| 05:56.44 | titter | Yup |
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| 05:56.58 | Micc | add a little delay on the shortel side if you can. |
| 05:57.27 | Micc | can you dial other things on the shoretel side from asterisk without a problem? |
| 05:57.37 | titter | Yup |
| 05:57.39 | titter | exten => _X.,1,Dial(SIP/shoresip/${EXTEN},45,r) |
| 05:57.42 | titter | thats all it does |
| 05:57.56 | titter | Let me pull the log from my laptop, 1 sec |
| 05:58.09 | Micc | I would try to put some kind of delay on the shoretel side. |
| 05:59.51 | Micc | I've gotta get some sleep before I pass out at my computer. |
| 06:00.18 | Micc | i know this is early for me, but I got up early this morning. |
| 06:00.40 | Micc | so good luck with that problem. good night |
| 06:03.12 | titter | http://pastebin.com/scP1zJY6 |
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| 06:16.18 | b14ck | sup |
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| 06:21.17 | ChannelZ | nuthin |
| 06:22.18 | b14ck | is doing some Asterisk coding for work tonight. |
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| 06:38.26 | Alton35 | Still no joy with chan_local. I will get back to screwing with it in a couple of days. |
| 06:41.22 | b14ck | Alton35, what's the problem with it? |
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| 06:43.44 | Alton35 | I want to dial out _within_ my AGI, |
| 06:43.57 | Alton35 | it's been a bit of a challenge. |
| 06:44.21 | Alton35 | Right now I seem to be able to invoke 2 cooperating AGIs, but the audio doesn't come through yet. |
| 06:44.49 | Alton35 | I wish I could just invoke one, but it seems that you have to invoke one to do the dialing and another to deal with the call after it connects. I have no idea. |
| 06:45.24 | b14ck | I'm confused. |
| 06:45.25 | b14ck | What do you mean? |
| 06:45.36 | Alton35 | well, think of a calling card program, |
| 06:45.36 | b14ck | If you post your directory structure, agi code, etc, i'll help |
| 06:45.49 | b14ck | You can dial out from within an agi |
| 06:45.53 | b14ck | just use exec |
| 06:45.59 | Alton35 | someone calls in, then you can dial out in your programs (my agi) and no problem, I have done a lot of that, |
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| 06:46.19 | Alton35 | but just to originate a call, sheesh, very difficult it seems. |
| 06:46.31 | b14ck | You want to originate a call from within an AGI? |
| 06:46.34 | b14ck | Or from nothing? |
| 06:46.39 | b14ck | (eg: no inbound call to trigger it) |
| 06:46.52 | Alton35 | right, strictly from nothing |
| 06:47.21 | Alton35 | actually a background process runs and monitors databases, then originates calls. |
| 06:47.35 | b14ck | ok, i can help with that |
| 06:47.40 | SiNGLer | is there a big difference betweek chan_h323 and chan_ooh323? |
| 06:47.42 | Alton35 | so with the .call file format I just get success or no success |
| 06:47.44 | b14ck | is the background process running on your asterisk server, or a remote server? |
| 06:47.48 | Alton35 | locally |
| 06:47.53 | b14ck | Alton35, use call files then |
| 06:47.58 | Alton35 | I do. |
| 06:48.02 | b14ck | what call files are you generating currently? (give me an example) |
| 06:48.11 | Alton35 | lemme make a new pastebin, just a minute |
| 06:48.12 | b14ck | I'm a call file pro: http://pycall.org/ |
| 06:48.22 | Alton35 | interesting |
| 06:48.23 | b14ck | SiNGLer, no idea |
| 06:49.13 | b14ck | What language are you using btw? |
| 06:49.21 | Alton35 | php + phpagi |
| 06:49.24 | b14ck | I'm pretty good with C, python, and PHP if your code is in there |
| 06:49.30 | b14ck | I can probably debug / figure out what the issue is. |
| 06:49.36 | b14ck | if you also include relevant code |
| 06:49.41 | Alton35 | sure, hold |
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| 06:54.52 | Alton35 | http://pastebin.com/WDAt4fCF |
| 06:55.22 | Alton35 | In this one I'm doing what I described, trying to use 2 cooperating PHP programs, one to do the dialing and another to do the logic after the call is answered. |
| 06:55.38 | Alton35 | I wish it could all be together like it should be. The last advice I got was that those functions should be separated. |
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| 06:57.11 | b14ck | your test call file has a problem in it |
| 06:57.23 | b14ck | see how in your channel line you have /n? |
| 06:57.31 | b14ck | was that supposed to be a '\n' (newline character)? |
| 06:57.42 | Alton35 | no, you're supposed to put that in |
| 06:57.57 | Alton35 | it has to do with chan_local |
| 06:58.07 | b14ck | Try do do: |
| 06:58.07 | Alton35 | something about not dropping the channel variables when connecting |
| 06:58.13 | b14ck | No. |
| 06:58.16 | b14ck | It will work properly withou tit. |
| 06:58.21 | b14ck | *without it |
| 06:58.22 | b14ck | =p |
| 06:58.53 | Alton35 | well, I haven't noticed any difference yet, but I figured that might be due to my own ineptitude... or poor .call file documentation! |
| 06:59.06 | b14ck | =p |
| 06:59.09 | b14ck | the documentation is poor |
| 06:59.14 | b14ck | i wrote some good docs on it if you're interested |
| 06:59.32 | Alton35 | interesting |
| 06:59.34 | b14ck | pycall.org has some, and my personal website has others: http://projectb14ck.org/ |
| 06:59.52 | b14ck | But anyhow, what happens when you spool one of those callfiles? (in the cli)? |
| 07:00.03 | Alton35 | ok, duly noted |
| 07:00.04 | b14ck | (make sure you do: core set verbose 99 before spooling it) |
| 07:00.12 | Alton35 | I have verbose 10 |
| 07:00.25 | b14ck | raise it |
| 07:00.27 | Alton35 | um, the system dials me |
| 07:00.31 | Alton35 | I didn't know it could go higher |
| 07:00.34 | b14ck | yeah |
| 07:00.36 | Alton35 | weird |
| 07:00.37 | b14ck | Also |
| 07:00.39 | Alton35 | hah |
| 07:00.41 | b14ck | do: core set debug 99 as well |
| 07:00.47 | Alton35 | same there, I thought it was 10 |
| 07:00.48 | b14ck | That will give you detailed information about the callfile as it's being processed |
| 07:00.51 | Alton35 | ok |
| 07:00.53 | Alton35 | great |
| 07:00.59 | Alton35 | anyway, it will call me, |
| 07:01.11 | Alton35 | then go silent, sorry, I should have included the CLI output, |
| 07:01.21 | b14ck | So it can connect to your extension 1000? |
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| 07:01.34 | Alton35 | anyway, when I hang up, the 2nd program then "answers" and starts into the call logic. |
| 07:01.48 | Alton35 | sure |
| 07:02.01 | b14ck | Well, the way it should work (based on your code), is: |
| 07:02.05 | b14ck | 1. It'll call your extension 1000 |
| 07:02.21 | b14ck | 2. Once you pick up, it'll immediately start executing announcements-calls-new,s,1 |
| 07:02.47 | b14ck | So, is it doing both those things? |
| 07:02.51 | b14ck | Or is it failing on an AGI? |
| 07:03.07 | b14ck | like announcement-2 or w/e |
| 07:03.09 | Alton35 | it doesn't seem to run step 2 until I hang up. |
| 07:03.27 | Alton35 | I thought that calling answer() in the program would connect the channels properly. |
| 07:03.46 | Alton35 | oops, sorry, hold on, |
| 07:03.47 | b14ck | Asterisk automatically answers() the call when it is set via the channel: or context: extensions |
| 07:04.03 | b14ck | so forcing an answer is redundant |
| 07:04.25 | Alton35 | well, I've had it be significant when trying to do silence detection |
| 07:04.28 | Alton35 | kinda odd, but anyway |
| 07:04.37 | b14ck | You can do a wait() instead |
| 07:04.39 | Alton35 | one question, |
| 07:04.44 | Alton35 | yes, wondered about taht |
| 07:04.47 | b14ck | Actually. |
| 07:04.54 | b14ck | There's an asterisk bug if youre doing silence detection. |
| 07:04.58 | Alton35 | my question is, can I do it all within one php program like it should be. |
| 07:05.05 | b14ck | If you don't play audio on the channel BEFORE detecting silence, it wont work |
| 07:05.16 | Alton35 | Yes, I learned that, and worked around that. |
| 07:05.20 | b14ck | ah =) |
| 07:05.23 | Alton35 | I can do that in the agi I assume, so no problem. |
| 07:05.25 | b14ck | writing a dialer? :) |
| 07:05.31 | Alton35 | I just want my bleepin' dial statuses. |
| 07:05.46 | Alton35 | The system calls to schedule people for openings, nothing spam-like |
| 07:05.56 | Alton35 | but I want my dial statuses like I got in my calling card program, |
| 07:06.06 | Alton35 | and I want to roll over if one provider is congested, etc. |
| 07:06.12 | Alton35 | the normal stuff anyone would want. |
| 07:06.26 | b14ck | That stuff should be handled by your system, not the call file code. |
| 07:06.34 | Alton35 | That's what I mean. |
| 07:06.39 | b14ck | eg: Instead of dialing through a AGI or whatever. |
| 07:06.41 | Alton35 | Can I get it into the php program where it should be? |
| 07:06.45 | b14ck | You should have a global subroutine that handles it |
| 07:06.48 | b14ck | In asterisk |
| 07:06.57 | b14ck | To fail over to trunks based on preference. |
| 07:07.13 | Alton35 | you mean in extensions.cofn |
| 07:07.15 | Alton35 | conf |
| 07:07.18 | b14ck | yep |
| 07:07.24 | b14ck | most people create an `outbound` context |
| 07:07.33 | b14ck | which does LCR as well as failover |
| 07:07.34 | Alton35 | but if I could just get the call I could do it all in my php program |
| 07:07.55 | b14ck | Want to paste your announcement-2.php code? |
| 07:07.59 | Alton35 | I know, I just want things to be more programmatically/database controlled. |
| 07:08.00 | b14ck | Probably getting stuck there. |
| 07:08.04 | Alton35 | ok |
| 07:08.18 | b14ck | Since you never get the call to [announcement-calls-new] |
| 07:08.51 | Alton35 | hmm, that paste was -2, lemme paste -1 here |
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| 07:10.00 | Alton35 | ok, http://pastebin.com/Aht7J4S9 |
| 07:10.18 | Alton35 | this is the "logic" side of it, what I really want to accomplish after I connect with the destination |
| 07:10.31 | Alton35 | and these were both in the same program, you can see the stuff commented out I think. |
| 07:10.40 | Alton35 | and I'd prefer them back that way if possible. |
| 07:11.21 | b14ck | also, try changing your contexts to this: http://pastie.org/1101629 |
| 07:12.28 | b14ck | So, when announcement-2.php is running, does it print 'announcement-1.php starting'? |
| 07:12.55 | b14ck | Also, if you want them in the same program, just accept command line arguments. |
| 07:13.03 | Alton35 | oh God, I stopped working on this the night before last |
| 07:13.04 | Alton35 | hmm |
| 07:13.06 | b14ck | EG: you could do: AGI(announcements.php,send) |
| 07:13.11 | Alton35 | anyway, trying to remember the details |
| 07:13.14 | b14ck | AGI(annoucements.php,receive) |
| 07:13.15 | b14ck | or whatever |
| 07:13.25 | b14ck | that's the same as doing: php announcements.php receive on the cli |
| 07:13.31 | b14ck | so just parse argv |
| 07:13.35 | b14ck | and run logic based on that |
| 07:13.36 | Alton35 | hmm, no possibility of putting it all back together, the program dials, the other end answers, we then continue to the logic? |
| 07:14.07 | Alton35 | It was so decent in the calling card program, I'd like to see that functionality again. |
| 07:14.09 | b14ck | Yah, you can do that. |
| 07:14.20 | b14ck | The php needs to be re-written. |
| 07:14.24 | Alton35 | I just don't see how. |
| 07:14.36 | b14ck | Here's some pseudo code (I'll pastebin it) |
| 07:14.43 | b14ck | just so you get the idea of what i mean |
| 07:15.10 | Alton35 | ok |
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| 08:14.38 | clekis | hello |
| 08:14.58 | clekis | is somebody in here, who can help me with T.38? |
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| 08:37.48 | ixyd | hi guys, is it possible to send the "ringing" line identification via P-asserted-identity in the ringing response to the phone from the asterisk server? |
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| 08:45.15 | Diffen2 | Hello, I have two asterisk server and the second one are connected to the first one over a registered sip-trunk. I guess i have missed something out here, but shouldnt my user account on the second asterisk |
| 08:45.41 | Diffen2 | be registered in some kind towards the first asterisk user extensions? |
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| 09:33.36 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-beta3 (2010/08/10), 1.6.2.11 (2010/08/10), 1.4.35 (2010/08/10), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
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| 09:41.56 | benedict | Hi @ all! Ive got a problem. Sometimes, when i dial a number i cannot connect to this number and i hear an announcement "Sorry" if i dial the number again everything is fine.. unfourtunately this error is not repuduceable, how can i debug it over a longer time? |
| 09:42.44 | fenrus | enable debugging and dial until you get the error |
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| 09:43.17 | fenrus | do you have that kind of playback configured somewhere in your pbx?, perhaps when some trunk is congested ? |
| 09:44.14 | benedict | it is a problem of one of our customers and i didnt setup this asterisk system, so i dont really know :( |
| 09:45.02 | benedict | how can i find out? |
| 09:45.40 | benedict | i have to say that im new to linux.. |
| 09:45.42 | fenrus | check the dialplan ? |
| 09:46.51 | benedict | ok |
| 09:47.27 | benedict | can you tell me how i can activate the debugging especially for my problem? |
| 09:49.35 | Tim_Toady | benedict this sounds like a freepbx setup |
| 09:50.02 | fenrus | no, since i dont know anything about your setup |
| 09:50.03 | fenrus | get someone that knows asterisk to have a look at it |
| 09:50.04 | Tim_Toady | propably some peer or phone is getting offline |
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| 09:50.49 | Tim_Toady | anyway log into aserisk console with 'asterisk -rvvv' and watchout for the messages while making the call |
| 09:51.20 | fenrus | that sounds really awsome on a high traffic astersik ;) |
| 09:51.32 | Tim_Toady | lol |
| 09:52.01 | benedict | yes it is freepbx, its not high traffic, i dont want to be connected to the console all the time so i wanne know how i can get the messages into a file |
| 09:52.02 | Tim_Toady | a grep like option on the console would be cool :P |
| 09:52.38 | Tim_Toady | benedict u can setup this in /etc/asterisk/logger.conf |
| 09:53.21 | Tim_Toady | i think its allready setup lke this if u run freepbx, check for /var/log/asterisk/full |
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| 09:56.06 | EmleyMoor | It even works without any "insecure" now - no explanation as to why |
| 09:56.12 | benedict | ok lets see.. |
| 09:56.38 | benedict | this file is 341mb.. |
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| 09:57.22 | Tim_Toady | benedict maybe u have to disable full logging when u re donr with that, or enable log rotation |
| 09:57.39 | Tim_Toady | s/donr/done/ |
| 09:58.13 | fenrus | lol |
| 09:58.18 | fenrus | runs away |
| 09:59.38 | benedict | does "full" also log the output that i normally get in the cli? |
| 10:00.37 | Tim_Toady | depends on the settings in logger.conf, in ur case i think yes, u get full logging of the console and debug messages |
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| 10:01.51 | benedict | in the logger.conf there is the entry set "full => notice,warning,error,debug,verbose" |
| 10:02.29 | Tim_Toady | it cant get more full that that :P |
| 10:02.42 | benedict | ok^^ |
| 10:03.38 | benedict | sorry for my stupid questions but i switched from windows to linux...and in windows everthing is so easy :/ |
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| 10:05.45 | Tim_Toady | to monitor in realtime without using the console just run 'tail -f /var/log/asterisk/full' |
| 10:06.45 | benedict | cool thank you |
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| 10:24.22 | russ | thats neat |
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| 10:25.00 | russ | the mmx code in the oslec echo canceling kernel module breaks the userspace codec_g729.so on my box |
| 10:25.16 | russ | compile the dahdi stuff without MMX, and everything is fine |
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| 10:29.04 | Tim_Toady | amd cpu russ? |
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| 10:33.04 | jayvee | I just Ctrl+Fâed my copy of Asterisk: The Future of Telephony, and there isnât one single mention of Jingle. Not only that, but there are zero Google search results for the error Iâm getting (âjingle_alloc: no jingle capable clients to talk to.â). Where could I get help on integrating Jingle and Asterisk? |
| 10:33.46 | jayvee | Ideally Iâd like to be able to both call Jingle clients from SIP, as well as SIP numbers from Jingle. But even if only one direction is working, it will open up a whole window of usefulness. |
| 10:34.41 | russ | Tim_Toady, via c7 |
| 10:34.47 | Tim_Toady | ah |
| 10:35.04 | russ | the mmx register saving code seems questionable |
| 10:35.05 | OlafsenM | guys, who is here running SS7 with FS? |
| 10:35.52 | russ | http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=593438 |
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| 11:00.36 | tzafrir | russ, hi |
| 11:01.19 | tzafrir | yes, got that report. looks odd. We do use oslec with VIAs |
| 11:04.12 | russ | there have been changes with the way fpu's and task swtich is handled in recent kernels afaik |
| 11:05.39 | tzafrir | I'm not sure I can get a working g729 codec. What other asterisk code can use MMX code? I suspect this is something to do with the same code on the stack |
| 11:06.00 | tzafrir | speex can use SSE2, IIRC but not mmx. What about gsm? |
| 11:06.05 | russ | you can probably compile any codec with mmx optimization |
| 11:06.40 | russ | but the g729 licensing would give you a go ahead to use it for testing/debugging iirc |
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| 11:08.15 | russ | the voip-info.org page claims 'Under patent law, it is a legitimate use to study or experiment with a patented technology without paying for a patent license' fwiw |
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| 11:09.06 | russ | you could probably reproduce with mmx code running at the same time as dahdi using the oslec |
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| 11:12.27 | russ | I at least know it would be bad if the kernel tried to do its own fpu operations inside a dahdi_kernel_fpu_begin/end block |
| 11:13.09 | russ | it seems to assume that it is in interrupt context, and several other bits of code in the kernel seem to assume that the fpu is ok to save/restore if it is not in interrupt context |
| 11:13.19 | russ | but not all the dahdi_kernel_fpu_begin/end blocks run in interrupt context |
| 11:14.09 | russ | but the oops I'm getting seems like it is from an NULL task struct or fpu struct |
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| 11:16.48 | E-bola | http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout gives a n error 400 |
| 11:16.55 | E-bola | its linked from http://www.asterisk.org/node/51413 |
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| 12:42.25 | Micc | what does Connected Party Identification Support mean? |
| 12:42.46 | wikii | IAx2 show Channels shows all active channels but their Format is "Unknown" |
| 12:42.47 | E-bola | There apear to be an issue with the contrib init.d script for debian in 1.8 beta3 |
| 12:42.59 | [TK]D-Fender | Micc: when the remote side gives you THEIR callerid when you call them |
| 12:42.59 | wikii | plese chk this link |
| 12:43.01 | wikii | http://ja.pastebin.ca/1920014 |
| 12:43.37 | Micc | nice. I assume that only works with other sip providers that suppor that? |
| 12:44.08 | [TK]D-Fender | Micc: I haven't heard of its use with ITSPs, jsut with PBXs for when you call an "internal extension" |
| 12:44.10 | [TK]D-Fender | ~cpid |
| 12:44.11 | infobot | [~cpid] Called-Party ID is possible with * using patches on Mantis. See : http://bugs.digium.com/view.php?id=8824 |
| 12:44.12 | [TK]D-Fender | ^^^ |
| 12:44.32 | Micc | oh ok |
| 12:45.03 | Micc | do most phones support that too? |
| 12:45.27 | wikii | ..@<[TK]D-Fender> plese check this link :: http://ja.pastebin.ca/1920014 |
| 12:45.42 | Micc | I don't see why I couldn't use it for in network calls too. |
| 12:46.34 | Micc | so what is Call Completion Supplementary Services support, and Advice of Change support? |
| 12:46.56 | Micc | I wish they put links to some of this stuff in the release notes. |
| 12:47.21 | [TK]D-Fender | wikii: USELESS... show the ENTIRE FAILED CALL |
| 12:47.28 | wikii | ok |
| 12:47.43 | [TK]D-Fender | Micc: I believe what you were mentioning was Advice Of Charge |
| 12:47.47 | birchquickly | Micc, these are standard terms you can Google.. I don't think the release notes are meant as a tutorial |
| 12:47.51 | Micc | wait its Advice of Charge support, yeah |
| 12:48.04 | Micc | right right, will do. |
| 12:48.11 | [TK]D-Fender | Micc: And I'm not sure about the meaning of the former |
| 12:48.56 | birchquickly | http://www.venturevoip.com/news.php?rssid=2359 |
| 12:49.05 | birchquickly | There's CCSS |
| 12:49.15 | Micc | I'm a bit excited about srtp. |
| 12:49.51 | Micc | that could be a new product offering for us. |
| 12:50.04 | birchquickly | SRTP is overrated |
| 12:50.17 | Micc | well is it secure or not? |
| 12:50.44 | Micc | why is it over rated to you? |
| 12:50.52 | birchquickly | Yes, so is wrapping your kid in bubble wrap before he goes to school. Overconcerned? Maybe. |
| 12:51.02 | coppice | it is secure if used properly. support is still patchu, though |
| 12:51.19 | coppice | VoIP without SRTP is crazy |
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| 12:54.58 | birchquickly | Do not love without a glove? |
| 12:55.50 | seanbright | E-bola: what issue would that be? |
| 12:58.09 | wikii | ..@<[TK]D-Fender> plese check this link :: http://ja.pastebin.ca/1920016 |
| 12:58.11 | Micc | this has got to be the most undefined unknown acronyms i've seen in a while. http://en.wikipedia.org/wiki/Advice_of_Charge |
| 12:58.29 | wikii | i hve uploaded all logs |
| 12:58.33 | Micc | I think I get the idea though, its for pay as you go cell phones mostly I think. |
| 12:59.32 | E-bola | seanbright: the part involving /etc/default/asterisk makes it not work for me. It just stops returning nothing, if i remove that part it works fine |
| 12:59.43 | fenrus | "pay as you go"? |
| 12:59.52 | fenrus | prepaid ? |
| 13:00.00 | [TK]D-Fender | wikii: Using a call-file to send yourself a fax? |
| 13:00.29 | Micc | birchquickly, I agree with you for the most part except when it comes to companies with unique scenarios that really do need to keep information under wraps. |
| 13:00.32 | [TK]D-Fender | wikii: nope... humm... |
| 13:00.54 | wikii | sorry cant understand ur question |
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| 13:01.47 | [TK]D-Fender | wikii: http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php |
| 13:01.49 | birchquickly | Micc: The odds of someone getting in place to intercept the RTP stream are so minimal it's not even worth arguing about. There's bigger things to worry about, IMHO |
| 13:01.51 | seanbright | E-bola: will take a look |
| 13:02.00 | [TK]D-Fender | wikii: Cause No. 28 - invalid number format (address incomplete). This cause indicates that the called party cannot be reached because the called party number is not in a valid format or is not complete. |
| 13:02.13 | [TK]D-Fender | wikii: - Channel 0/1, span 1 got hangup, cause 28 <----------- |
| 13:02.19 | seanbright | E-bola: i don't have /etc/default/asterisk and it is working fine for me |
| 13:02.29 | Micc | birchquickly, I know and I agree, unless you know your being gunned for. |
| 13:02.40 | wikii | but when i call the same number from my sip phone it rings |
| 13:02.41 | wikii | :( |
| 13:02.54 | E-bola | seanbright: i tried to remove it as well, didnt change anything, was quite weird.... |
| 13:02.56 | [TK]D-Fender | wikii: I don't see anything usable to compare |
| 13:02.56 | seanbright | E-bola: do you get an error message or it just silently fails? |
| 13:03.03 | E-bola | the only weird thing on the system is that the kernel is relatively old |
| 13:03.08 | E-bola | seanbright: silently fails |
| 13:03.17 | seanbright | E-bola: hmm |
| 13:03.21 | WIMPy | wikii: A 'type of number' problem? |
| 13:03.23 | Micc | birchquickly, if someone wants to get in the middle its not that hard. |
| 13:03.24 | E-bola | it quits in the if sentence where its supposed to check for the default file |
| 13:03.27 | wikii | tell me whhich logs you need i will upload them |
| 13:03.36 | E-bola | i traced itwith simple echo sentences to see where it stopped |
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| 13:04.01 | [TK]D-Fender | wikii: I don't see the COMPLETE failed call (no PRI DEBUG in there), and I don't see this GOOD call you said this should mirror. |
| 13:04.01 | seanbright | E-bola: and /etc/default/asterisk doesn't exist? |
| 13:04.14 | E-bola | seanbright: nope, i delled it |
| 13:04.22 | E-bola | makes no sence at all.... |
| 13:04.36 | seanbright | tzafrir: any idea on E-bola's issue? |
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| 13:04.44 | seanbright | master of all things debian |
| 13:04.50 | Micc | birchquickly, although its much more difficult than the old days of tapping a land line. |
| 13:05.07 | wikii | how i can debug pri?? tell me command |
| 13:05.19 | SiNGLer | E-bola: debian use that default file for settings and for asterisk enabling |
| 13:05.46 | E-bola | seanbright: you know if the default file is new? i dont normally use it when compiling asterisk myself, so im wondering if it was used in earlier asterisk versions of the init script |
| 13:05.50 | SiNGLer | where should be line, which you modify/uncomment/whatever to allow asterisk startup |
| 13:05.56 | WIMPy | wikii: Sorry just came in and didn't get the whole story. |
| 13:06.03 | [TK]D-Fender | wikii: "pri debug span 1" |
| 13:06.28 | SiNGLer | E-bola: modify startup script :) afaik debian package use default file (even 1.4) |
| 13:06.57 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 13:07.04 | wikii | ok thankyou.. i will upload all logs tomorw and let you know.. rite now iam away from server acess |
| 13:07.40 | E-bola | SiNGLer: yes the debian packages, but this is the tarball |
| 13:07.58 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 13:08.06 | wikii | WIMPY .... whenevr i send fax "Busy Signal is detected" |
| 13:08.09 | SiNGLer | and you use debian startup script, which probably is from package, dunno :) |
| 13:08.26 | wikii | Wiphy...http://ja.pastebin.ca/1920016 |
| 13:08.34 | SiNGLer | solution would be to use blank default, or modify script |
| 13:08.38 | E-bola | no i dont, im sory but i guess you dont know the tarball too well. There are included init scripts for the common dists, including debian |
| 13:08.43 | SiNGLer | I'd go for script modification |
| 13:08.59 | E-bola | I already have a solution, i just removed the problem parts of the init script. Im just reporting back to be nice :) |
| 13:09.24 | SiNGLer | E-bola: I know tarball, probably into tarball was included script, which was used for debian package |
| 13:09.51 | WIMPy | wikii: You send out a PRI? And you get an address incomplete wehn fending fax but not when calling the same number from a phone? |
| 13:10.10 | E-bola | SiNGLer: if i recall correctly the init scripts are different |
| 13:11.00 | wikii | yeah i send it on pri |
| 13:11.10 | wikii | yeah true |
| 13:12.41 | [TK]D-Fender | wikii: "busy signal detected" is NOT the problem |
| 13:12.50 | [TK]D-Fender | wikii: You call did NOT go through. |
| 13:13.37 | WIMPy | And a lot of obfuscation. |
| 13:13.43 | wikii | ok <[TK]D-Fender> ... where iam doing wrong.. in configuration ?? |
| 13:14.00 | [TK]D-Fender | wikii: Who says the CONFIGS are at fault? |
| 13:14.10 | wikii | iam asking sir |
| 13:14.14 | [TK]D-Fender | wikii: You can't even look at 2 calls right now, so just come back later. |
| 13:14.22 | wikii | i am new t asterisk |
| 13:14.39 | [TK]D-Fender | wikii: Don't ask for an autopsy when you can't even present the dead body |
| 13:14.56 | fauxalliance | i thought i smelled something fetid |
| 13:15.03 | wikii | <[TK]D-Fender> sorry . :( |
| 13:15.32 | WIMPy | wikii: Apart from that number problem you also have a BC problem. SPEECH is going to cause trouble when sending faxes. |
| 13:15.59 | wikii | ok |
| 13:16.03 | [TK]D-Fender | WIMPy: there is no signal for "fax"... it is not a "data" call |
| 13:16.25 | WIMPy | But a pri intensive debug of both a successfull and a failed call will help. |
| 13:16.30 | [TK]D-Fender | wikii: Forget that last remark and come back with debug of a failed and a matching good call as you have claimed. |
| 13:16.46 | WIMPy | [TK]D-Fender: It's 3.1kHz Audio for G3 Fax. |
| 13:17.13 | wikii | OK sure Thankyou both.. thanks for your time |
| 13:17.39 | WIMPy | Many PKXs won't route a speech call to a fax. |
| 13:17.56 | WIMPy | s/PKX/PBX/ |
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| 13:21.12 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:32.23 | tzafrir | E-bola, still here? |
| 13:32.37 | tzafrir | I see that this script will fail if readlink is not installed |
| 13:33.13 | benedict | hi! I have an 'Asterisk 1.4.29.1-BRIstuffed-0.4.0-RC3l' and 'console dial' does not exist, how can i make an outgoing call over the cli with that version? |
| 13:35.04 | grolloj | Hi. Is it possible on an asterisk host to use one IP on the host for all SIP traffic and a second IP for all RTP traffic? |
| 13:35.39 | tzafrir | benedict, originate? |
| 13:35.54 | tzafrir | Do you want to generate a call from the sound card? |
| 13:35.59 | Naikrovek | core show application originate |
| 13:36.04 | Naikrovek | i think |
| 13:36.25 | tzafrir | If so, you probably need something along the lines of: module load chan_alsa.so |
| 13:36.32 | tzafrir | or: module load chan_oss.so |
| 13:36.34 | benedict | i only wanna check the connection and make an outgoing call to my mobile phone or sth |
| 13:36.35 | tzafrir | first |
| 13:36.47 | E-bola | tzafrir: yep |
| 13:36.59 | benedict | its not necessary to hear sth |
| 13:37.23 | E-bola | i dont have a readlink package on this debian server.... |
| 13:37.47 | E-bola | i have /lib/init/readlink |
| 13:39.13 | [TK]D-Fender | grolloj: No. * is not a "SIP Server". it is a B2BUA |
| 13:41.21 | grolloj | hmm. ok. asterisk would still be handling both the media and signaling. i thought i might be missing something in a nat config. |
| 13:41.41 | grolloj | basically, trying to keep sip on a private net and open rtp up public. |
| 13:42.35 | [TK]D-Fender | grolloj: There is no split. |
| 13:43.07 | grolloj | fair enough. thanks. |
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| 13:51.08 | tzafrir | E-bola, but it is not in the PATH |
| 13:51.15 | tzafrir | right? |
| 13:51.34 | tzafrir | anyway, what happens if you run the script? nothing? |
| 13:51.36 | E-bola | well is /lib/init/readlink the readlink u refer to? |
| 13:51.42 | E-bola | precisely, nothing |
| 13:57.16 | benedict | Ive another question to originate, i don`t get it.. I want to make a call to 01577xxxx in context 'test' via zap... Hows the syntax? |
| 13:58.00 | [TK]D-Fender | benedict: {channel] is as you you would put in a Dial() comamnd |
| 14:01.56 | benedict | so i have to make 'originate Zap/g1/01577xxx' ? |
| 14:02.06 | muiro | When using queues, I believe I remember being able to designate a local channel to ring, but then some other device to use for status. Is my memory off or can I do that? |
| 14:02.29 | seanbright | muiro: you can do that |
| 14:03.27 | seanbright | from queues.conf.sample: |
| 14:03.32 | seanbright | member => Local/1000@default,0,John Smith,SIP/1000 |
| 14:04.11 | muiro | ah, okay, I was just opening that up to read it |
| 14:04.12 | muiro | thanks much |
| 14:04.33 | muiro | though, one followup. Can I use custom device states for status? |
| 14:05.00 | seanbright | not sure. |
| 14:05.04 | muiro | it's easy to test |
| 14:05.09 | muiro | I'll report back |
| 14:06.25 | *** join/#asterisk bougyman (bougyman@pdpc/supporter/gold/bougyman) |
| 14:08.29 | benedict | no i executed: originate Zap/01577xxx@test extension 01577xxx@test but nothing happend :( |
| 14:09.32 | benedict | *now |
| 14:10.02 | [TK]D-Fender | benedict: Do"Zap/01577xxx@test" look like something valid to put in a Dial() to you? |
| 14:11.33 | tzafrir | E-bola, so try: sh -x /etc/init.d/asterisk start |
| 14:12.05 | *** join/#asterisk md0 (~mdavid@c-98-222-7-129.hsd1.il.comcast.net) |
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| 14:14.07 | stope | I just installed asterisk from source but app_mysql is not there, what am I forgetting? |
| 14:14.18 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 14:14.33 | anonymouz666 | app_mysql really sucks |
| 14:15.23 | stope | ok, well what do you suggest? |
| 14:16.11 | stope | It may suck in your opinion dude but it is relatively robust |
| 14:16.28 | anonymouz666 | Use func_odbc instead |
| 14:16.39 | benedict | [TK]D-Fender: Oooopa^^ You`re right, i forgot the 'g1' -.-* |
| 14:18.45 | [TK]D-Fender | benedict: "@" has no place is a ZAP CALL |
| 14:20.22 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:20.52 | benedict | yes sorry, was a typo 'originate Zap/g1/01577xxx extension 01577xxx@extension_custom.conf' works |
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| 14:22.39 | [TK]D-Fender | benedict: @extension_custom.conf' <- pardon? You have a CONTEXT that looks like a FILE NAME? |
| 14:24.31 | benedict | yes, but i works :S @test works too |
| 14:26.39 | benedict | confusing, but both works... its only for testing, so it doenst matter^^ |
| 14:27.18 | *** join/#asterisk festr_ (~festr@nostromo.flh.cz) |
| 14:27.26 | festr_ | hi, can DAHDI sniff T1 to pcap? |
| 14:27.34 | festr_ | or it only does wanpipe? |
| 14:27.38 | festr_ | (sangoma) |
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| 14:35.40 | *** join/#asterisk ixti (~ixti@217.70.249.69) |
| 14:35.56 | ixti | Hello, everybody. |
| 14:36.12 | ixti | Can somebody help with asterisk 1.6.2 + gafachi ? |
| 14:36.52 | ixti | After they switched to new clusters I can't get it to work (old config works but not very reliable, while new config does not works at all) |
| 14:37.28 | Kyosh | new cluster? |
| 14:37.56 | ixti | Here's what I receive in console with sip set debug on : http://pastebin.com/vSDSpG6F |
| 14:38.16 | ixti | Kyosh: well, I just cited their website ;)) |
| 14:38.47 | jamko | ixti: I use gafachi with 1.6 .. when did problems start? |
| 14:38.48 | seanbright | pastebin.com is giving me adds for asiandating.com |
| 14:38.57 | seanbright | targeted advertising is AMAZING |
| 14:38.59 | seanbright | heh |
| 14:39.01 | jamko | seanbright: yay! |
| 14:39.33 | Kyosh | ixti, dont trust provider to give valid working examples of configs |
| 14:40.49 | Kyosh | its up to you to know how to config |
| 14:40.52 | Kyosh | with that said |
| 14:41.02 | ixti | jamko: well, before they added gafachi1a, gafachi1b there was only one gafachi. so when I left old server - it works (sometimes) when i put new one (e.g. 1A (67.216.35.162) Rochester, NY SIP Cluster) it doesn't |
| 14:41.24 | jamko | I terminate to Rochester. What is the issue exactly? |
| 14:41.26 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 14:41.53 | ixti | Kyosh: well, with old server same config was working good |
| 14:42.00 | Kyosh | oy their examples are a headache |
| 14:42.05 | ixti | jamko: it just doesn't calls |
| 14:42.13 | Kyosh | oh you mean the cluster of servers |
| 14:42.19 | Kyosh | go back to the old config |
| 14:42.21 | drmessano | seanbright: Lucky you.. all my pastebin ads want me to buy livestock |
| 14:42.27 | drmessano | seanbright: :( |
| 14:42.37 | Kyosh | drmessano must be in the stix |
| 14:42.56 | ixti | Kyosh: and that's the funniest part - it doesn't work with new servers too :)) |
| 14:42.59 | Kyosh | go back to the old config, change the ip's to match gafachi 1a or 1b |
| 14:43.16 | Kyosh | know what |
| 14:43.17 | Kyosh | nm |
| 14:43.18 | *** join/#asterisk darkskiez (~dz@62-50-207-34.client.stsn.net) |
| 14:43.29 | Kyosh | loves his targeted asian girl ads |
| 14:43.49 | ixti | one second I'll try to write it from scratch. |
| 14:43.54 | jamko | ixti: are you ip authenticating? hopefully |
| 14:44.36 | jamko | gettin' me some asian. |
| 14:45.32 | ixti | jamko: nope |
| 14:45.37 | ixti | jamko: should i? |
| 14:46.27 | jamko | I don't see why not. |
| 14:47.14 | jamko | ixti: ip auth - easy, easy like sunday morning. |
| 14:47.37 | ixti | well, basically looks like authentication work just fine. at least I see gafachi1a/username 67.216.35.162 in the sip show peers |
| 14:48.29 | *** join/#asterisk modsaid (~modsaid@82.201.210.162) |
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| 14:48.46 | ixti | but i can't make a call. will try to setup ip authenticating now. |
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| 14:51.12 | ixti | no difference :(( |
| 14:51.28 | jamko | Did you setup ip auth at the gafachi site? |
| 14:51.35 | jamko | you have to put your ip address into their gui. |
| 14:53.32 | ixti | jamko: yes. I know. and I have did it |
| 14:59.17 | E-bola | Hmm 1.8 beta3 keeps giving me this when testing 1 snom320 to another: No SRTP module loaded, can't setup SRTP session. |
| 14:59.45 | E-bola | guess i gotta disable rtp encryption on the phones |
| 15:02.34 | jamko | ixti: what is the message on the cli when the call bounces |
| 15:02.55 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:05.11 | ixti | jamko: -- Attempting call on SIP/01134617179433@gafachi for s@pstn-incoming:1 (Retry 1) |
| 15:05.18 | *** join/#asterisk coppice (~chatzilla@m121-203-203-90.smartone-vodafone.com) |
| 15:05.29 | ixti | jamko: and then == Using SIP RTP CoS mark 5 and that's all :(( |
| 15:05.45 | ixti | verbose and debug are 3 |
| 15:08.07 | ixti | jamko: ahhh... i'm stupid idiot.... |
| 15:08.17 | ixti | sorry... |
| 15:09.21 | ixti | was trying to make a call on SIP/...@gafachi instead of SIP/...@gafachi1a |
| 15:10.09 | ixti | jamko: does ip based authentication really better than standard one/ |
| 15:10.11 | ixti | ? |
| 15:12.23 | *** join/#asterisk dailylinux (~test@s21-00210.dsl.no.powertech.net) |
| 15:13.01 | *** join/#asterisk mnuzaihan (~tohyttym@bb116-14-149-15.singnet.com.sg) |
| 15:13.04 | mnuzaihan | Hi |
| 15:13.15 | mnuzaihan | i am having problem with "HangUp()" |
| 15:13.41 | mnuzaihan | It seems that after i had recorded my voicemail and press pound key (#), it doesn't hangup. |
| 15:14.15 | mnuzaihan | even though i have "HangUp()" at the end of my extension configuration |
| 15:14.45 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:15.12 | Kobaz | mnicholson: more info needed.... where is the call coming from... and going to (what technology.. .analog/t1/sip/etc) |
| 15:15.48 | mnuzaihan | the call is coming from my client -> asterisk server on public IP |
| 15:15.54 | mnuzaihan | asterisk using SIP |
| 15:19.42 | mnuzaihan | After i press pound and got a "goodbye" audio, the session is still active. |
| 15:20.09 | mnuzaihan | The problem might be due to this: |
| 15:20.10 | mnuzaihan | <PROTECTED> |
| 15:20.10 | mnuzaihan | <PROTECTED> |
| 15:20.27 | mnuzaihan | The Hangup must be after "Spawn Extension" if i'm right |
| 15:20.28 | ruyo | Does _*XX*X.# match? |
| 15:20.29 | ruyo | Considering I need a number that can vary length between * and #. |
| 15:21.11 | *** join/#asterisk bobboau (~bobboau@65.87.32.252) |
| 15:21.36 | bobboau | is anyone awake in here? |
| 15:22.48 | ixti | thanks everybody. :)) |
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| 15:45.07 | ChannelZ | nooooope |
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| 16:05.58 | p3nguin | ruyo: I think _*XX*X.# will match one * followed by two numbers followed by one * followed by one number and another one or more characters followed by # |
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| 16:06.22 | nny | Odd situation, I have a setup that has worked fine for a couple of years, and suddenly has started someting weird. I have a specific subnet attached to a seperate switch using a VLAN, and all of a sudden they can dial out, but dialing in just terminates. Pastebin here, I am the only one with access to any of the switches |
| 16:06.22 | nny | http://pastebin.org/614249 |
| 16:07.36 | ruyo | p3nguin, that's what I'm aiming for. I'll be able to test it in a while and I'll let you know. |
| 16:08.01 | *** join/#asterisk Faithful (~Faithful@nat65.mia.three.co.uk) |
| 16:08.21 | [TK]D-Fender | [12:05]<p3nguin>ruyo: I think _*XX*X.# will match one * followed by two numbers followed by one * followed by one number and another one or more characters followed by # <- no |
| 16:08.31 | p3nguin | If you only need two numbers between the two * it might work. |
| 16:08.49 | hrhrhr | how can i check if the number dialled by the user is only 6 numbers, then append an std code? |
| 16:10.23 | hrhrhr | ,70,twW) <--- and what does twW mean |
| 16:10.36 | hrhrhr | t=timeout |
| 16:10.37 | hrhrhr | wW? |
| 16:10.54 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 16:11.10 | [TK]D-Fender | hrhrhr: "core show function LEN" |
| 16:11.22 | [TK]D-Fender | hrhrhr: "core show application dial" |
| 16:11.36 | hrhrhr | teach a man to fish, eh? :P |
| 16:11.45 | [TK]D-Fender | hrhrhr: and "t" != timeout |
| 16:12.12 | [TK]D-Fender | heads to lunch |
| 16:16.01 | *** join/#asterisk mechbangirc (~mechbangi@mbl-65-157-238.dsl.net.pk) |
| 16:16.13 | mechbangirc | hi channel |
| 16:17.36 | mechbangirc | i just installed asterisk and zaptel packages on debian lenny. when i try to connect to asterisk 'asterisk -vr' i get this error in logs "res_config_pgsql.c:782 pgsql_reconnect: Postgresql RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info." any suggestion? |
| 16:18.17 | mechbangirc | i have mysql cdr enabled in cdr.conf i dont know why postgresql is there, i want to use mysql |
| 16:18.45 | *** join/#asterisk ruyo (~psantos@a83-132-248-161.cpe.netcabo.pt) |
| 16:19.08 | ruyo | [TK]D-Fender, p3nguin, it did work. |
| 16:21.00 | fenrus | disable postgresql then |
| 16:21.02 | ruyo | With *XX*X.*X.# I can make, for instance *67*12345*543# |
| 16:21.17 | Corydon76-dig | mechbangirc: When you install from packages, you need to seek support from the package maintainer, first |
| 16:22.12 | *** join/#asterisk adyn (~adyn@unaffiliated/adyn) |
| 16:23.23 | mechbangirc | Corydon76-dig: i dont know much about package management, anyway thanks |
| 16:24.33 | Corydon76-dig | mechbangirc: any idea what version it is? |
| 16:25.31 | mechbangirc | 1.4.21.2 |
| 16:26.07 | Corydon76-dig | mechbangirc: that's tremendously old |
| 16:27.11 | mechbangirc | actually i want to use a2billing, and guys at a2billing say either 1.4 or 1.2. |
| 16:27.26 | Corydon76-dig | 1.2 is on security support only |
| 16:28.14 | Corydon76-dig | But I mean that 1.4.21.2 is a tremendously old version of 1.4. We're now at 1.4.35 |
| 16:28.47 | mechbangirc | i ve used 1.6 for quite some time. now for a2billing i have to downgrade. I always used to compile asterisk. this time i installed from package and now i dont know what is happening |
| 16:29.07 | mechbangirc | yea debian loves old stuff |
| 16:29.20 | russ | is it one of the built in vertical service codes interfering? |
| 16:29.33 | Corydon76-dig | Debian also loves installing poorly tested patches in their packages |
| 16:29.50 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 16:29.55 | bougyman | explain poorly tested? |
| 16:30.19 | bougyman | you may be talking about sid or squeeze. stable is tested by CI, users, security team, others. |
| 16:30.39 | mechbangirc | i dont know much about debian. before that i had centos it was great but now i like archlinux more than any distro |
| 16:30.43 | Corydon76-dig | bougyman: your mysql odbc driver is utter crap |
| 16:31.04 | gr0mit | ok, anyone recommend a good DID provider for a tollfree number? |
| 16:31.10 | bougyman | gr0mit: flowroute |
| 16:31.28 | Corydon76-dig | bougyman: I had to put a workaround in Asterisk for the crap that is the packaged odbc driver |
| 16:31.31 | gr0mit | US tollfree i mean |
| 16:31.33 | drmessano | I second Flowroute |
| 16:31.50 | gr0mit | ok i will google them |
| 16:32.08 | Corydon76-dig | bougyman: to be fair, it's not just Debian, but package maintainers in general that add patches that get rejected upstream |
| 16:32.29 | bougyman | Corydon76-dig: is your bug http://bugs.debian.org/cgi-bin/pkgreport.cgi?src=myodbc there? |
| 16:32.32 | mechbangirc | ok now unload(ed) => problematic.modules and finally there is only one warning left "pbx.c:2981 ast_register_application: Already have an application 'Directory'" |
| 16:32.57 | Corydon76-dig | bougyman: nope |
| 16:33.05 | mechbangirc | anyone? |
| 16:33.13 | bougyman | Corydon76-dig: if it were you'd be part of the solution. |
| 16:33.19 | bougyman | they don't let grave bugs sit for long. |
| 16:33.24 | Corydon76-dig | bougyman: tis easier for me to install those from source. And faster. |
| 16:33.55 | bougyman | can you explain the env so I can replicate it? |
| 16:34.01 | bougyman | i never use mysql, so haven't run into it. |
| 16:34.43 | Corydon76-dig | bougyman: shared connection, connection times out, first thread reconnects and everything works fine. Second thread attempts to use the reconnected connection and the process crashes |
| 16:35.11 | bougyman | shared connection meaning two asterisks? |
| 16:35.22 | bougyman | or two things using the unixODBC data source together? |
| 16:35.29 | Corydon76-dig | shared connection means multiple threads in a single process |
| 16:36.27 | Corydon76-dig | I added idletimeout in res_odbc to workaround this specific problem |
| 16:36.55 | Corydon76-dig | If we drop the connection ourselves and reconnect (as opposed to the server timing out the connection), no problem |
| 16:37.11 | mechbangirc | i am now going to compile 1.4.35. i noticed 1.8.0 beta 3 is out |
| 16:37.29 | *** join/#asterisk JamesVoss (j@unaffiliated/jamesvoss) |
| 16:37.53 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 16:37.55 | bougyman | doesn't * have native mysql support without using odbc, Corydon76-dig ? |
| 16:38.05 | bougyman | reading newsgroup threads about this |
| 16:38.12 | Micc_ | I'm starting to think orderlyq is going out of business or they just have really bad IT people. |
| 16:38.13 | jamko | bougyman: yes |
| 16:38.54 | jamko | odbc I believe is only needed for func_odbc, or with integration into shared db with opensips. |
| 16:38.58 | Corydon76-dig | bougyman: How do you think I test ODBC? |
| 16:39.08 | jamko | or other 3rd party solutions |
| 16:39.13 | bougyman | Micc_: how's that? |
| 16:39.22 | bougyman | we use orderlyq, it's been great for our collection managers. |
| 16:39.47 | bougyman | it's just a template for us (development) so we can replace it with out own, but it's bridging the gap between our old proprietary one and our new internally devved one. |
| 16:40.16 | Corydon76-dig | And yes, I nearly forgot about func_odbc |
| 16:40.46 | bougyman | Corydon76-dig: understood. i know some of that packages maintainers, i'll see if I can replicate it and get it fixed, if it's not already fixed in squeeze. |
| 16:41.11 | *** join/#asterisk riddlebox (~james@p54895264.dip.t-dialin.net) |
| 16:42.37 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 16:43.59 | Micc_ | bougyman, when I signed up for a trial, I got an error that the device was out of space. So then after I sent them an email about it a few days later they fix it and I was able to create a test account. Now I can't login because the server seems to be offline. the main site is there but after login it redirects to us3.orderlyq.com |
| 16:44.25 | bougyman | Micc_: oh, we use orderlystatsSE, a self-hosted version. |
| 16:44.29 | bougyman | i never tried their hosted. |
| 16:46.12 | gr0mit | Micc_ what is the prb with orderlyq? |
| 16:49.46 | [TK]D-Fender | [12:21]<ruyo>With *XX*X.*X.# I can make, for instance *67*12345*543# <- nothing after that "." matters |
| 16:50.58 | *** join/#asterisk Justman (~just@justerr.kgn.ru) |
| 16:51.03 | ruyo | [TK]D-Fender, indeed. |
| 16:51.51 | bougyman | i detest the pattern matching. |
| 16:52.12 | bougyman | why not just give people pcre or posix regex? |
| 16:52.15 | ruyo | In that case I can only have *XX*XX*X.. It's a good enough match. |
| 16:52.49 | Micc_ | gr0mit, us3.orderlyq.com is down. |
| 16:53.21 | gr0mit | eeew |
| 16:53.35 | bougyman | this is why I didn't use the hosted product. |
| 16:54.47 | hrhrhr | if i pass a call from pbx 1, to pbx2, over iax, can i send it out of pbx2's zap channel, using 's' exten to pass the call to dial (with EXTEN)? |
| 16:55.07 | hrhrhr | which bit of the book should i read for this |
| 16:55.16 | drmessano | All of it |
| 16:55.33 | hrhrhr | it wont dial unless i have a dialplan for the specific number |
| 16:55.36 | [TK]D-Fender | hrhrhr: Every call is jsut a call. From a phone to *. Out to another *. In on that other *. Out to something else. |
| 16:55.39 | Micc_ | most hosted companies are on top of that stuff like white on rice. |
| 16:55.46 | hrhrhr | i think 's' should let me get away with this but it doesn't work |
| 16:55.51 | bougyman | Micc_: i haven't noticed such. |
| 16:55.57 | [TK]D-Fender | hrhrhr: and the "s" exten isn't magical. Doesn't matter what yuo use for patterns as long as it leads to the outcome you want |
| 16:55.59 | bougyman | i've seen outages with some of the largest. |
| 16:56.15 | bougyman | heck I recovered amazon to a backup site in jersey one time, they were completely incompetent. |
| 16:56.17 | [TK]D-Fender | hrhrhr: using 's' exten <-- dosn't actually eman anything |
| 16:56.20 | [TK]D-Fender | mean* |
| 16:56.30 | ruyo | hrhrhr, if you Dial(Zap/1(${EXTEN}) from the 's' extension, you'll Dial(Zap/1/s) |
| 16:56.33 | hrhrhr | i've included a context, top of the list which processes s |
| 16:56.43 | gr0mit | have you called orderlyq? |
| 16:56.44 | hrhrhr | oh |
| 16:56.59 | hrhrhr | right |
| 16:57.10 | Micc_ | I know outages happen, but not knowing about a major outage till a customer tells you is not acceptable for a hosted company in my mind. |
| 16:57.24 | Micc_ | yeah, their overseas number isn't working for me. |
| 16:57.50 | drmessano | Micc_, My biggest peeve is when the company blames their hosting or upstream and takes NO blame on themselves |
| 16:58.02 | gr0mit | Micc they are based in UK |
| 16:58.09 | gr0mit | I met the guy who runs it |
| 16:58.12 | *** join/#asterisk hfb (~hfb@96.247.66.242) |
| 16:58.37 | drmessano | Micc_, THEY are the ones who chose their provider and their hosting, they are the ones reselling/selling services using the aforementioned.. they should take some credit by proxy |
| 16:58.43 | Micc_ | I know. I'm not sure why the number wasn't working, maybe it was a vitelity problem, I'll try calling again. |
| 16:58.54 | gr0mit | try calling their UK number |
| 16:59.18 | Micc_ | gr0mit, thats what I meant, I tried their UK number. |
| 16:59.25 | drmessano | Maybe they are down |
| 16:59.26 | gr0mit | Call us on +44 845 0045 413 |
| 16:59.31 | gr0mit | this one? |
| 17:00.01 | Micc_ | yup |
| 17:00.03 | gr0mit | i get straight through |
| 17:00.19 | gr0mit | must be your telco |
| 17:00.25 | Micc_ | busy. |
| 17:01.00 | [TK]D-Fender | hrhrhr: extension doesn't matter as long as it leads to what you want it to do. |
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| 17:01.27 | p3nguin | I've had issues calling UK numbers on VoIP.ms (who is using Vitelity) in the past several days. |
| 17:02.53 | p3nguin | Just fyi. |
| 17:07.38 | *** join/#asterisk morex (~m@host86-178-202-206.range86-178.btcentralplus.com) |
| 17:07.40 | morex | Hi there |
| 17:07.57 | morex | I gather there have been some problems reported here with the US OrderlyStats managed service |
| 17:08.08 | gr0mit | p3nguin, what numbers are you having issues with? |
| 17:08.11 | morex | I do technical support for Orderly Software who provide the service |
| 17:08.16 | Micc_ | I opened a trouble ticket with vitelity about it a few days ago, they just said to try it again. |
| 17:08.36 | morex | We are having an issue with one of our US servers, us3, which has developed a slow-fail on the hard drive |
| 17:08.45 | Micc_ | morex, yes, I'm having problem loging into my trial account on us3.orderlyq.com/members |
| 17:08.55 | Micc_ | Can you move my account to another server? |
| 17:08.58 | *** join/#asterisk sol (~sol@unaffiliated/sol) |
| 17:09.33 | morex | We've already ordered a replacement server from our hosting provider |
| 17:09.49 | p3nguin | gr0mit: 0845 072 7227, 0845 071 0759, and the 845 0045 413 that you pasted. |
| 17:09.50 | morex | And we've moved the hard-drive intensive processes to a different server |
| 17:10.08 | morex | So there shouldn't be any further problems moving forwards today, and we hope to have a full replacement tomorrow |
| 17:10.16 | morex | I am monitoring the server closely |
| 17:10.41 | russ | I can't call 011448450045413 with teliax |
| 17:10.49 | gr0mit | p3nguin, these are all UK non-geo numbers |
| 17:10.59 | gr0mit | with higher than standard call rates |
| 17:11.07 | gr0mit | wonder if they are being blocked? |
| 17:11.19 | morex | Micc: Yes your account will be moved to the new server as soon as it's available |
| 17:11.52 | morex | If you have any further difficulty, please call us straight away on +448450045413 or +442075827228 |
| 17:12.02 | morex | It does look like this will be the last of the problems today though. |
| 17:12.28 | gr0mit | morex, they had issues calling your 0845 number from outside UK |
| 17:12.37 | gr0mit | it was me that called you just now |
| 17:12.46 | gr0mit | as they could not call the 0845 number |
| 17:13.12 | morex | Ah yes sometimes the carrier blocks it |
| 17:13.20 | morex | I'll add a geo number to our web site now. |
| 17:13.26 | gr0mit | good plan! |
| 17:13.34 | gr0mit | hates 0845 numbers |
| 17:13.38 | Micc_ | morex, can't you move my account to a server that is working right now? |
| 17:13.50 | Micc_ | or is it only on that one server |
| 17:14.13 | Justman | Hey guys. Are there any callbacks mechanisms in Asterisk able to raise events on hangup while ast_structure containing my dialplan variables is still accessible. Thanks in advance for any help. |
| 17:15.09 | morex | It's just this one server that's affected by the hardware issue |
| 17:15.18 | morex | And it is working right now |
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| 17:34.27 | bmoraca_work | 01189998819991197253 |
| 17:35.19 | *** join/#asterisk mnuzaihan (~tohyttym@bb116-14-149-15.singnet.com.sg) |
| 17:36.39 | mnuzaihan | hi all. I am using asterisk v1.6 (SIP) and whenever i use time or voicemail (after pressing pound to finish with message), the connection is not terminated and from ngrep it doesn't show any BYE. |
| 17:37.32 | mnuzaihan | i keep seeing "INVITE" though |
| 17:40.35 | [TK]D-Fender | mnPerhaps you should be showing us the failed call |
| 17:40.40 | [TK]D-Fender | mnuzaihan: Perhaps you should be showing us the failed call |
| 17:42.22 | mnuzaihan | [TK]D-Fend: the calls did not fail. But at the end of everything (time, voicemail records and pressing #), the session is still active and i did not see any "BYE" from asterisk. |
| 17:42.31 | mnuzaihan | i'm using SIP |
| 17:48.00 | drmessano | So there's nothing wrong with the call? |
| 17:50.08 | carrar | never say goodbye |
| 17:50.22 | carrar | bon jovi does it so well |
| 17:52.40 | mnuzaihan | drmessano: It cannot hangup automatically after pressing pound in voicemail, or after reading on the time/date. |
| 17:53.05 | mnuzaihan | drmessano: the session is still active and it doesn't hang up by itself. |
| 17:53.07 | drmessano | Ok, so the call FAILED, and you were asked to pastebin a debug |
| 17:53.44 | carrar | Can you move the microphone a little bit closer please |
| 17:56.05 | fenrus | will the real slim shady please stand up |
| 17:57.36 | drmessano | I think he's trying to make a wish on an airplane |
| 17:57.37 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 17:57.49 | [TK]D-Fender | [13:50]<carrar>never say goodbye <- I play that with my band |
| 17:58.05 | [TK]D-Fender | carrar: And jammed it up yesterday with another group I play with |
| 17:58.16 | drmessano | Stick to your Guns is the best Bon Jovi song, ever. |
| 17:59.13 | mnuzaihan | drmessano: http://pastebin.com/epVA4nLF |
| 17:59.21 | drmessano | Funny that they perform it so little, there's a video of them trying to do it live, and Jon is holding a lyric sheet.. and still messed up the song |
| 17:59.30 | [TK]D-Fender | drmessano: I did "Wild Is The Wind" at an acoustic open-mic night last week... |
| 17:59.42 | drmessano | Ah, love that one too |
| 17:59.57 | *** join/#asterisk mechbangirc (~mechbangi@mbl-65-157-238.dsl.net.pk) |
| 18:00.12 | [TK]D-Fender | mnuzaihan: ASTERISK SIP DEBUG FROM CLI + VERBOSE |
| 18:00.16 | mnuzaihan | drmessano: those are the ones that i see in ngrep. no "bye" statements and after it announces the time, the connection is not dropped. |
| 18:00.19 | drmessano | I remember the day I bought New Jersey, when I lived in New Jersey, better than I remember 911 |
| 18:00.24 | [TK]D-Fender | mnuzaihan: Clear enough? We want PROOF. Complete circunstances |
| 18:00.26 | mnuzaihan | TK: ok |
| 18:00.59 | drmessano | I call it BJNJ Day |
| 18:01.54 | mechbangirc | i setup res_fax and res_fax_digium modules, checked the status from cli>fax show stats everything looks ok. so what next! in the dialplan how do i make use of it. btw this is free one line licence from digium. |
| 18:02.23 | mnuzaihan | [TK]D-Fend: http://pastebin.com/NGxfiHkS |
| 18:03.01 | *** join/#asterisk vgster (~vgster@94-194-190-189.zone8.bethere.co.uk) |
| 18:05.25 | [TK]D-Fender | mnuzaihan: see every transmission failing. I'd go check your FIREWALLS if I were you |
| 18:05.39 | *** part/#asterisk Justman (~just@justerr.kgn.ru) |
| 18:06.46 | mnuzaihan | [TK]D-Fend: running off VPN with a separate range of VPN client IP than LAN, so i guess my connection is NAT-ed. |
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| 18:10.32 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 18:10.58 | [TK]D-Fender | mnuzaihan: I guess you need to completely reevaluate your networking. |
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| 18:17.41 | carrar | [TK]D-Fender, lets hear some sound tracks!! |
| 18:18.13 | [TK]D-Fender | carrar: You on FB? |
| 18:18.17 | carrar | hell no |
| 18:20.08 | drmessano | I was gonna add you, carrar |
| 18:20.09 | drmessano | :( |
| 18:20.16 | *** join/#asterisk fofware (~fabian@186.125.127.156) |
| 18:20.39 | carrar | I have my own web site, no need to maintain two sites |
| 18:20.49 | drmessano | You have the awkward social deficiencies of an Amateur Radio operator or somethin? |
| 18:20.54 | carrar | and turn over my privacy control to some company |
| 18:21.04 | carrar | haha |
| 18:21.21 | [TK]D-Fender | carrar: I have nothing personal on mine. And I'm only tagged in 4 videos from my band's page. |
| 18:21.48 | carrar | FB is just more thing to waste my time on checking |
| 18:21.58 | [TK]D-Fender | Full privacy block, no personal data, no photos, no useless apps, etc. |
| 18:22.01 | carrar | however I do bgp peer with facebook |
| 18:22.05 | carrar | but thats it! |
| 18:22.44 | [TK]D-Fender | carrar: Well as a musician this IS a good means for me to keep up with events I do actually care about. That's about it. No family members, ex's, etc on there. Just closer friends and music contacts to keep me up to date |
| 18:24.17 | carrar | Aren't there any muscians version of FB? |
| 18:24.28 | carrar | There is a maoney making oppertunity |
| 18:24.31 | carrar | (c) |
| 18:25.37 | [TK]D-Fender | carrar: Actually there is.. one where you can do track mixing as well so you can collaborate with oterh musicians on a fixed recording,e tc |
| 18:25.54 | [TK]D-Fender | carrar: Forgot the name, but I'm sure its easy to find |
| 18:26.23 | drmessano | I wonder if they used anything from Justin Frankel's Ninjam project |
| 18:26.47 | drmessano | That seemed like a pretty awesome platform for collaboration |
| 18:27.29 | *** join/#asterisk mboehn (mathias@mboehn.alfa-skk.pr0jectX.net) |
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| 18:51.44 | Qwell | [TK]D-Fender: who's the guy that works at/for CAT? |
| 18:51.49 | Qwell | happen to know offhand? |
| 18:52.15 | [TK]D-Fender | No idea |
| 18:54.20 | seanbright | bob? |
| 18:57.32 | Qwell | Naikrovek does. |
| 18:58.36 | seanbright | i was close |
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| 19:06.54 | mnuzaihan | [TK]D-Fender: I've remotely accessed the machine on LAN via RDP (there is no firewall between the remotely-accessed-client and server) and i have the same problem. |
| 19:09.55 | [TK]D-Fender | mnYour packets aren't geting responded to... nothing much we can say here... |
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| 19:25.15 | raden | Qwell, its Naikrovek hes gone for day headache |
| 19:25.23 | raden | Qwell, something I can help you with |
| 19:25.35 | Qwell | is that english? |
| 19:25.57 | raden | Qwell, is what English ? |
| 19:27.44 | raden | yawn , I'm outta here |
| 19:28.40 | Qwell | raden: I mean I don't understand your response. |
| 19:35.12 | raden | Qwell, Yes Naikrovek is the one who works at cat , he is gone today left work early with a headache . |
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| 19:38.30 | Qwell | I see |
| 19:43.56 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
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| 20:08.01 | Kyosh | does asterisk support UUI? |
| 20:09.55 | *** join/#asterisk defsdoor (~andy@plingit.gotadsl.co.uk) |
| 20:11.47 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 20:18.05 | pabelanger | Kyosh: UUI? Avaya? |
| 20:32.32 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 20:47.00 | *** join/#asterisk fofware (~fabian@186.125.124.113) |
| 20:49.17 | Kyosh | hell if i know. im not even sure what UUI is |
| 20:55.58 | ChannelZ | User User Interface! So nice they named it twice |
| 20:59.21 | drmessano | Wait, you want to know if Asterisk supports UUI and you don't know what UUI is? |
| 20:59.28 | drmessano | Yes, no, yes |
| 21:05.12 | ChannelZ | Urge Urinary Incontinence |
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| 21:31.28 | Kyosh | drmessano, someone asked me and i didnt know how to answer. so i said "hell if i know", but then the boss said "find out" |
| 21:31.30 | Kyosh | so here i am |
| 21:31.39 | Kyosh | not like its my job, but the boss says it is |
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| 21:42.43 | *** join/#asterisk nny (~Scott@65.23.110.106.nw.nuvox.net) |
| 21:42.46 | nny | ~book |
| 21:42.47 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:43.20 | *** part/#asterisk lukhas (~lucas@bearstech/lukhas) |
| 21:45.50 | ChannelZ | Well it's not like it's our job either |
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| 21:53.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:56.39 | golikwid|mac | hey fellas |
| 21:58.07 | ChannelZ | heeeeyy sexay ladyy |
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| 22:18.38 | jtodd | Kyosh: Asterisk supported UUI information elements on Q.931, but it is unknown if they currently work. |
| 22:19.05 | jtodd | Kyosh: They were utilized for experimentation by AT&T testing some years back. Search the mailing lists. |
| 22:21.30 | seanbright | there was a #define in chan_dahdi to enable it; not sure if it's still there |
| 22:22.11 | seanbright | /* define this to send PRI user-user information elements */ |
| 22:22.17 | seanbright | #undef SUPPORT_USERUSER |
| 22:22.38 | seanbright | that's in sig_pri.c in trunk. in chan_dahdi.c in older branches i'd guess. |
| 22:27.11 | jtodd | seanbright: Thanks, that's it. |
| 22:27.55 | jtodd | Not sure if the original asker is still here, but that's the answer. Now, how it works, EXACTLY, is up to speculation. Most networks don't transit it over network borders, and some don't even move it aorund internally. |
| 22:28.12 | jtodd | Or it's different length limits based on what type of switch is connecting. Not very standardized. |
| 22:28.26 | jtodd | Best To Try It Yourself. BTTIY. |
| 22:34.26 | *** join/#asterisk tessier (~treed@kernel-panic/copilotco) |
| 22:36.28 | tessier | Anyone ever have a problem with only one Aastra phone at a time being able to register out of 6 phones NATing out through a Linux/Shorewall firewall to an asterisk box behind another Linux/Shorewall firewall behind a one-to-one NAT (so port numbers and everything map correctly)? |
| 22:36.51 | tessier | For some reason it is just this one location. Other locations with phones behind NAT all talk to asterisk perfectly. |
| 22:37.09 | jamko | NAT issue. |
| 22:37.11 | tessier | What is odd about this location that does not work is that the phone system used to be local to these phones. |
| 22:37.22 | tessier | jamko: Yes. I'm wondering exactly what the issue is though. |
| 22:37.31 | jamko | NAT : ) |
| 22:37.51 | tessier | Other remote locations behind NAT (Linux based wrt54g's) are working perfectly. |
| 22:37.53 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 22:38.04 | tessier | So it seems it would have to be a NAT issue local to these non-working (except for one) phones. |
| 22:38.12 | tessier | I suspect a bogus setting in the phone somewhere. |
| 22:38.27 | tessier | nat=1 for all of these phones in sip.conf |
| 22:38.48 | jamko | I assume you are using separate sip ports for each phone. |
| 22:39.20 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 22:39.49 | tessier | jamko: No. |
| 22:39.58 | tessier | jamko: And neither are the correctly working remote phones behind NAT. |
| 22:40.12 | jamko | well that is a small miracle. |
| 22:40.16 | tessier | I've set up tons of asterisk and Snom phones. |
| 22:40.35 | tessier | And a few crap Grandstreams. And they have never needed such things. Nor does it make sense that they would be needed. |
| 22:40.43 | tessier | The main factor is having a decent NAT implementation. |
| 22:40.55 | tessier | All of the low end crap NAT boxes that cause those troubles greatly complicate things and I avoid them. |
| 22:41.00 | tessier | Linux on both ends for me. |
| 22:41.15 | jamko | ok... well good luck with that .. I suggest you use separate SIP ports. |
| 22:41.16 | tessier | But a TCP connection is a tuple of local and remote ports. |
| 22:41.25 | tessier | Well, port and IP. |
| 22:45.40 | jamko | How exactly is your wrt54g knowing which device to send traffic to, if you can only point 5060 to one ip address at a time? If the request originated behind your NAT, then you might have a chance, but if the request originates outside your NAT, good luck man. The solution is not a better NAT device, the solution is not to use NAT period. SIP and NAT don't mix, never will, and never did. |
| 22:47.28 | Nugget | NAT blows goats |
| 22:47.51 | jamko | But, best effort configuration if NAT is absolutely necessary, is to use separate SIP and RTP ports for each device behind the NAT. |
| 22:48.06 | jamko | period. |
| 22:48.47 | carrar | IPv6 4ALL |
| 22:48.56 | jamko | word up carrar! |
| 22:49.11 | carrar | w3rD Y0!! |
| 22:49.29 | carrar | I got my /32 |
| 22:50.00 | bougyman | a /32 is one ip isn't it? |
| 22:50.04 | bougyman | or are you talking ipv6? |
| 22:50.08 | carrar | IPv4 it is |
| 22:50.13 | carrar | but we're talking IPv6 |
| 22:50.18 | bougyman | ah cool. |
| 22:50.50 | *** join/#asterisk jetlag (~jetlag@pool-173-61-206-191.cmdnnj.east.verizon.net) |
| 22:51.20 | bougyman | tunnel broker or direct from ISP? |
| 22:51.32 | carrar | direct allocation from ARIN |
| 22:51.48 | bougyman | does your ISP offer v6? |
| 22:51.51 | carrar | yes |
| 22:51.58 | carrar | whois -h whois.radb.net \!6as7752 |
| 22:52.03 | bougyman | nice to know some will. |
| 22:52.11 | bougyman | i got on the beta program with the only isp here who offers it. |
| 22:52.17 | bougyman | they still haven't rolled it out. |
| 22:52.18 | bmoraca_work | yeah, a /32 is an "isp" increment...organizations are generally given /64s from ISPs |
| 22:52.30 | bougyman | i have two /48s |
| 22:52.39 | bmoraca_work | HE is giving those out |
| 22:52.56 | bougyman | but they're just toys without an isp that supports v6 |
| 22:53.19 | carrar | we have 4 full transite IPv6 providers |
| 22:53.31 | carrar | plus lots of peers |
| 22:53.57 | bmoraca_work | i got a /64 from HE when they first started doing it...but i never set it up |
| 22:53.59 | p3nguin | IPv6 is still a novelty in most places. |
| 22:54.11 | bougyman | that's how the telcos around here treat it. |
| 22:54.21 | bougyman | only a couple small ones are even considering it. |
| 22:54.22 | bmoraca_work | IPv6 is so overengineered that it likely won't be fully deployed for another 10 years |
| 22:54.24 | p3nguin | Until they NEED it, they will continue that way. |
| 22:54.39 | bougyman | and charter (the worst ISP here) is the only one offering it, and only in beta, and they're delayed on deployment. |
| 22:54.48 | carrar | lame |
| 22:55.04 | carrar | most people won't be prepaired |
| 22:55.07 | carrar | unfortunately |
| 22:55.24 | p3nguin | Charter is the worst in your area? |
| 22:55.26 | carrar | and people who are, will stand to gain |
| 22:55.38 | bougyman | p3nguin: for the most part. |
| 22:55.56 | bougyman | i'm sure there are some I haven't tried or had a client use that may be worse. |
| 22:56.10 | p3nguin | Charter is usually among the best. |
| 22:56.12 | bougyman | but couldn't be much worse than charter without a business death wish. |
| 22:56.17 | bougyman | they're the CI Host of ISPs. |
| 22:56.44 | carrar | the only reason for a ISP to not deploy it is lazyness |
| 22:57.18 | carrar | granted thats a great excuse |
| 22:57.39 | p3nguin | I like that Charter offers Ultra60. That's a nice speed for home cable modem service. |
| 22:59.29 | bmoraca_work | charter around here caps off at 10mbit, and if you pay for that, you're more likely to get about 5mbit |
| 23:11.50 | *** join/#asterisk Letoric (Letoric@253.sub-75-199-49.myvzw.com) |
| 23:12.17 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 23:12.34 | *** join/#asterisk lyetz (~lyetz@me.lyetz.me) |
| 23:12.52 | Letoric | hiyas. I'm having a bit of difficulty with asterisk and * when dialed from my handsets. It keeps reporting invalid extension even though the dialplan context has _*4101 as an extension |
| 23:12.58 | Letoric | any thoughts on what I'm screwing up? |
| 23:15.17 | Letoric | running 1.6.2.7 for the code |
| 23:17.03 | Letoric | hello? |
| 23:17.35 | WIMPy | Not unless you tell us what the problem is. But _*4101 does not look like a sensible extension. |
| 23:18.24 | Letoric | The error is 'Call from '4108' to extension '*' rejected because extension not found |
| 23:18.33 | Letoric | so it never lets me finish dialing anything past the * |
| 23:18.40 | Letoric | it instantly handles it as a call |
| 23:19.12 | Letoric | if I specifically put only * as an extension, it processes that ok |
| 23:19.31 | Letoric | (if I add that to the dial plan context) |
| 23:19.36 | raden | p3nguin, i have ultra 60 |
| 23:20.45 | raden | http://www.speedtest.net/result/921001394.png <<<< actual results |
| 23:21.28 | raden | I can get 30 - 35 MB burst sometimes |
| 23:21.50 | *** join/#asterisk lyetz (~lyetz@me.lyetz.me) |
| 23:23.00 | *** part/#asterisk nny (~Scott@65.23.110.106.nw.nuvox.net) |
| 23:27.38 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 23:29.50 | Letoric | any idea WIMPy? |
| 23:30.26 | Letoric | I have looked through the dial plan several times, and have even gone as far as purging everything in the context, to see if it would at least wait for me to press additional digits |
| 23:30.48 | Letoric | the only time it seems to allow me to use * is if I specifically make an extension for * by itself ;( |
| 23:31.39 | WIMPy | Why do you use pattern matching but have no pattern? |
| 23:32.02 | Letoric | I found an article for how to allow people to transfer directly to voicemail |
| 23:32.16 | Letoric | it suggested using _*${EXTEN} |
| 23:32.32 | Letoric | except it's not really the variable, just did that to make it easier to comprehend ;p |
| 23:33.04 | Letoric | I guess I can try it without the pattern match |
| 23:33.40 | Letoric | same issue |
| 23:33.57 | Letoric | before I can even attempt to press another key, it is already issuing busy signal |
| 23:34.30 | WIMPy | What kind of technology? Could it be the terminal? |
| 23:35.01 | Letoric | the phone? |
| 23:35.02 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 23:35.23 | WIMPy | yes |
| 23:35.36 | Letoric | the phone works ok on another asterisk system, running 1.4.x code base |
| 23:35.47 | Letoric | different dial plan there though, everything was slopped into 1 context |
| 23:36.11 | Letoric | or nearly everything at least |
| 23:36.15 | Letoric | users were broken out |
| 23:36.25 | Letoric | phones started in default though, which contained 99% of all code |
| 23:36.37 | golikwid|mac | can anyone think of a good few songs for a construction company's MOH |
| 23:36.40 | WIMPy | But it works with numbers starting with *? |
| 23:36.43 | Letoric | it is a Polycom Soundpoint IP 670 though |
| 23:36.44 | golikwid|mac | road construction |
| 23:36.45 | Letoric | yeah |
| 23:36.59 | WIMPy | golikwid|mac: Bob the builder? :-) |
| 23:37.04 | golikwid|mac | hm |
| 23:37.14 | Letoric | on the old system, it works. I was trying to start the new system from scratch and build it more in line with public perception of best practices |
| 23:37.26 | Letoric | that way it's easier for the next guy to walk in behind me ;> |
| 23:37.53 | TJNII | On the road again, Can't drive 55, "Damn this traffic jam" (or whatever its called by DaVinci's notebook, King of the road.... |
| 23:38.03 | golikwid|mac | Where the Blacktop ends, She Thinks my tractors sexy, bless this broken road, where the sidewalk ends, we built this city, road construction is what i have so far |
| 23:38.06 | TJNII | I'd go for traffic jam songs, but I can only think of one.... |
| 23:38.13 | golikwid|mac | i have king of the road too |
| 23:38.19 | golikwid|mac | love cant driv 55 |
| 23:38.23 | WIMPy | When you use a sip phone, it ususlly does not do overlap dial, that is it shouldn't contact Asterisk before you completed dialling. |
| 23:38.24 | golikwid|mac | and in the road again! |
| 23:38.56 | WIMPy | So I'd check the phones dialplan. |
| 23:39.09 | Letoric | I am, just not understanding what is catching it |
| 23:39.19 | Letoric | the console isn't providing much insight, nor is messages |
| 23:39.48 | WIMPy | Look at the phone, not the server. |
| 23:40.11 | Letoric | can you elaborate some? |
| 23:40.24 | Letoric | I'm rather new at this, not sure exactly what I'd be looking for on the phone |
| 23:40.33 | golikwid|mac | TJNII: i like the idea im concerned about the DAmn in the song though...conservatives might not like it |
| 23:40.37 | Letoric | I have about a week of experience with asterisk heh |
| 23:41.18 | jamko | letoric: What kind of phone? |
| 23:41.28 | Letoric | Polycom Soundpoint IP670 |
| 23:41.53 | golikwid|mac | it's a tele-phone |
| 23:42.01 | golikwid|mac | is that a kind? |
| 23:42.18 | jamko | lol |
| 23:42.59 | WIMPy | Letoric: Look for a dial paln on the phone. |
| 23:43.13 | WIMPy | s/paln/plan/ |
| 23:43.25 | jamko | im pushing f and 10 damn it.. |
| 23:43.30 | jamko | nothing happening. |
| 23:44.09 | golikwid|mac | are you looking at the settings through the web interface of the phone? |
| 23:44.15 | golikwid|mac | or the phones menu |
| 23:44.19 | jamko | letoric: ip hope you are using the config files |
| 23:44.56 | jamko | that's where the backbone of your polycom is.. web interface, not so much. |
| 23:45.01 | Letoric | I am |
| 23:45.07 | Letoric | the config files don't have any dial plan in them |
| 23:45.13 | golikwid|mac | but the web interface is prettier |
| 23:45.13 | Letoric | for the phone, that is |
| 23:45.33 | Letoric | don't even know the password to use the web interface of the phone heh |
| 23:45.42 | golikwid|mac | 654 Polycom |
| 23:45.44 | Letoric | just using tftp (don't yell!) to configure it |
| 23:45.50 | golikwid|mac | well Polycom 654 |
| 23:46.26 | Letoric | that didn't work ;( |
| 23:46.37 | golikwid|mac | what didt work |
| 23:47.07 | Letoric | polycom username, password 654 |
| 23:47.09 | golikwid|mac | username: Polycom Password: 654 or are you not talking to me |
| 23:47.11 | Letoric | or vice versa |
| 23:47.18 | golikwid|mac | capital P |
| 23:47.19 | Letoric | yeah, I was talking to you goliwid, thanks |
| 23:47.22 | Letoric | k |
| 23:47.34 | Letoric | no worky ;( |
| 23:47.36 | golikwid|mac | thats the default for polycom |
| 23:47.37 | golikwid|mac | hum |
| 23:47.45 | jamko | letoric: sip.cfg - dialmap section |
| 23:48.15 | golikwid|mac | opps |
| 23:48.16 | golikwid|mac | lol |
| 23:48.21 | golikwid|mac | its Polycom and 456 |
| 23:48.25 | golikwid|mac | im lisdexic |
| 23:49.08 | golikwid|mac | Letoric: did that work? |
| 23:49.36 | Letoric | checking, was looking in the cfg file |
| 23:49.47 | golikwid|mac | i dont deal with polycom much my clients are too cheap :( |
| 23:49.49 | Letoric | yes, that did work. Thanks |
| 23:50.02 | golikwid|mac | but they did give me one to try out so i have one on my desk lol |
| 23:50.45 | jamko | golikwid: what do you use? |
| 23:50.52 | golikwid|mac | Aastra |
| 23:50.57 | jamko | nm.. misread what you said.. |
| 23:51.01 | golikwid|mac | cause they are pretty |
| 23:51.10 | jamko | thought you said polycoms were cheap. |
| 23:51.13 | golikwid|mac | and i can brand them |
| 23:51.15 | golikwid|mac | no sir |
| 23:51.18 | jamko | purdy |
| 23:51.20 | golikwid|mac | my customers are though |
| 23:51.24 | jamko | yay |
| 23:51.55 | jamko | most customers are.... forest for the trees I tell em' ... no sir.. |
| 23:52.25 | *** join/#asterisk russ (~russ@206.29.188.182) |
| 23:52.30 | golikwid|mac | yea i have a doctors here that prefers paying me to clean up other peoples messes than just paying me in the first place |
| 23:52.43 | Letoric | the dialplan in sip.cfg seems pretty simple |
| 23:52.49 | Letoric | I don't see anything about * |
| 23:53.07 | Letoric | and the timeout is 3|3|3|3|3|3 |
| 23:53.07 | golikwid|mac | are you using the asterisk hunter |
| 23:53.16 | Letoric | not sure what the asterisk hunter is, sorry |
| 23:53.36 | golikwid|mac | its a commercial around here that one of the cable companies use |
| 23:53.40 | golikwid|mac | i think its Brighthouse |
| 23:53.41 | golikwid|mac | anyway |
| 23:53.43 | *** join/#asterisk russ (~russ@206.29.188.182) |
| 23:53.52 | golikwid|mac | it was alot funnier in my head |
| 23:53.58 | Letoric | heh |
| 23:54.12 | Letoric | we're going live with the new system tomorrow at lunch |
| 23:54.22 | golikwid|mac | you'll be fine |
| 23:54.24 | Letoric | I'm trying to work all the minor kinks out before then, this is really the only one plaguing me |
| 23:54.38 | golikwid|mac | as long as you can get asterisk to play the technical difficulties message your good |
| 23:54.46 | Letoric | hehe |
| 23:55.02 | golikwid|mac | the number you have dialed is not in servie |
| 23:55.09 | golikwid|mac | customers love that one |
| 23:55.56 | Letoric | so......any other thoughts on the issue with *? |
| 23:56.12 | golikwid|mac | whats it doing |
| 23:56.14 | golikwid|mac | or not doing |
| 23:56.20 | golikwid|mac | i didnt really see your orig problem |
| 23:56.28 | Letoric | haha |
| 23:56.41 | golikwid|mac | hey i gave you the web password |
| 23:56.45 | golikwid|mac | is helpfull |
| 23:57.24 | Letoric | <Letoric> The error is 'Call from '4108' to extension '*' rejected because extension not found |
| 23:57.24 | Letoric | <Letoric> so it never lets me finish dialing anything past the * |
| 23:57.24 | Letoric | <Letoric> it instantly handles it as a call |
| 23:57.24 | Letoric | <Letoric> if I specifically put only * as an extension, it processes that ok |
| 23:57.24 | Letoric | <Letoric> (if I add that to the dial plan context) |
| 23:57.40 | Letoric | The phone I'm using starts in context [phones] |
| 23:58.04 | Letoric | that context includes 3 other contexts, [internal] [parkedcalls] [outgoing_calls] |
| 23:58.21 | Letoric | internal includes a context for [stdexten] |
| 23:58.23 | golikwid|mac | from-internal? |
| 23:58.37 | Letoric | don't have a from-internal |
| 23:58.44 | golikwid|mac | hm |
| 23:59.07 | Letoric | if it's internal, and it's being dialed from a phone with context phones, I would think that the include would hand off ok, and it does for other extensions |
| 23:59.12 | Letoric | it's only * that is giving me hell |
| 23:59.18 | golikwid|mac | ok so the dialplan in the phone is matching it and sending it out |
| 23:59.33 | golikwid|mac | so what is your phones dialplan |
| 23:59.34 | Letoric | yes, the phone is sending it to asterisk, asterisk is rejecting it |
| 23:59.48 | golikwid|mac | well maybe its rejecting it cause its partial |
