| 00:00.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:00.21 | phunyguy | that way the PC on the other end of the modem just gets set up with DHCP and it grabs the public IP from your ISP |
| 00:00.38 | gregd | Chotaire: ok, so now my asterisk box is connected to my provider without any retransmits |
| 00:00.56 | Chotaire | gregd: can you ping whatever host on the internet from your asterisk box? |
| 00:00.56 | phunyguy | my ISP may be special though, they were setting us up for fiber, and now I just have a cat5 jack in here |
| 00:01.04 | gregd | yes |
| 00:01.13 | gregd | Chotaire: i can ping, anything and connect to any port |
| 00:01.16 | Chotaire | gregd: good.. now you need to get your sip client back up with a proper ip address. |
| 00:01.31 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 00:01.42 | Chotaire | you will need to find out if you hear audio transmissions and stuff.. |
| 00:02.39 | Chotaire | your sip client must have an ip in range 192.168.1.0/24 or 169.254.255.0/16 |
| 00:03.02 | Chotaire | if you don't wanna search for the cause of that stupid ip address lease, use a static ip for the time being. |
| 00:04.01 | gregd | Chotaire: one moment... got something strange again... my sip client is totally off and asterisk comes up with: http://pastebin.com/hFHPVKib |
| 00:05.02 | Chotaire | 172.16.128.1 |
| 00:05.04 | Chotaire | there we go again |
| 00:05.21 | gregd | Chotaire: but why the asterisk box send the messages out? |
| 00:05.59 | Chotaire | have you restarted asterisk in the meanwhile? and fixed the ip addresses of your sip client? |
| 00:07.09 | Chotaire | checked /var/spool/asterisk/outgoing ? |
| 00:07.09 | gregd | Chotaire: yes, i swithced off totally my sip client and restarted asterisk.. that is what it comes with |
| 00:07.09 | *** join/#asterisk digitalirony (~digitalir@my.ass.looks.just.like.your-face.info) |
| 00:07.09 | gregd | Chotaire: spool/asterisk/outgoing is empty |
| 00:07.20 | Chotaire | fix that macosx box first, give it a static proper ip address that's on the same subnet than the asterisk box and try again. |
| 00:07.45 | Chotaire | verify (route -n) that the other addresses are no longer present... |
| 00:07.58 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:08.06 | Chotaire | ehrm ifconfig/ip addr whatever. |
| 00:08.18 | Chotaire | I don't do darwin. |
| 00:09.33 | Chotaire | alteast during these tests this evil 172.16 private range needs to be eradicated in all connected devices (or these devices be disconnected) |
| 00:10.31 | gregd | Chotaire: i mean, i am not able to modify the darwin box (sip client).... but how does this should affect asterisk box... asterisk box is not aware about it AFAIK |
| 00:11.02 | Chotaire | NOTIFY sip:6000@172.16.128.1:64450 SIP/2.0 |
| 00:11.03 | gregd | though, i can use a different sip client on different box |
| 00:11.14 | Chotaire | yes please ;) let's give steve a rest... |
| 00:11.22 | gregd | ;) |
| 00:11.50 | gregd | i mean, i've got smartphone with built in simple sip client |
| 00:12.28 | Chotaire | if we keep on screwing around with that macosx box we might end up uninstalling adobe flash and you will need to change ISP to AT&T. |
| 00:13.11 | gregd | but the thing that worries me, at current stage, nothing is connected to the asterisk box (neitiher is trying to connect to it), but the asterisk box (freshly started) still issues the messages.. whare are they coming from? |
| 00:13.11 | Chotaire | why not try the smartphone, just make sure it will get an address within the desired range via dhcp. |
| 00:13.46 | Chotaire | cd /etc/asterisk ; grep -R 172.16. |
| 00:13.49 | Chotaire | just out of curiosity... |
| 00:13.59 | gregd | moment.. |
| 00:15.45 | p3nguin | Why would you need to cd in order to grep? |
| 00:15.56 | p3nguin | grep -r stuff /path/to/things |
| 00:16.08 | Chotaire | p3nguin: you're right. I always cd in first because I usually do more afterwards. |
| 00:16.25 | Chotaire | saves me lifetime. |
| 00:16.34 | gregd | Chotaire: there is no 172.16 in the whole asterisk folder... except of sip.conf where is it commented out |
| 00:16.57 | Chotaire | commented out by ";", right? |
| 00:17.01 | gregd | yes |
| 00:17.04 | Chotaire | k ;) |
| 00:17.35 | gregd | let me restart the whole asterisk box... will be back in 1 minutes |
| 00:17.40 | Chotaire | anyway, try to connect with whatever sip client, with 192.168.1.* ip, to your asterisk box and place an outgoing call. |
| 00:17.42 | Chotaire | k |
| 00:21.40 | *** join/#asterisk gregd_ (~gregd@cpc9-sgyl29-2-0-cust103.sgyl.cable.virginmedia.com) |
| 00:22.03 | gregd | still the same thing... |
| 00:25.37 | Chotaire | what would be 172.16.128.1? |
| 00:26.01 | Chotaire | ah, that's the phone. |
| 00:26.21 | gregd | lol.. |
| 00:26.59 | gregd | i forgot to remove milkfish before installing asterisk... it was stll running ;) let me try again |
| 00:27.14 | Chotaire | ok |
| 00:28.26 | gregd | 172.16 still there ... :/ |
| 00:28.38 | gregd | let me remove the user... |
| 00:28.49 | Chotaire | ip addr add 172.16.128.123 dev br0 ; ping 172.16.128.1 |
| 00:29.06 | Chotaire | if you can ping that host, you still did not get rid of that ip address on your sip client! |
| 00:29.52 | Chotaire | that would be ip addr add 172.16.128.123/24 dev br0 ..sorry. |
| 00:30.21 | gregd | ok.. problem solved.. i renamed user 6000 to 6002 in users.conf |
| 00:30.28 | gregd | the message went off :) |
| 00:30.44 | Chotaire | maybe cached somewhere? dunno. so... what's next. |
| 00:31.02 | gregd | so i've got a clean asterisk box that's connected to my sip provider... |
| 00:31.20 | gregd | how can i make sure that they can establish a call in between? |
| 00:31.26 | Chotaire | try an outgoing call and before you authenticate to asterisk, make sure there is no other 172.16.* addresses configured on the sip client's box. |
| 00:31.45 | gregd | how can i make an outgoing call without having client connected? |
| 00:32.01 | Chotaire | you better be using a client.. you want to test audio transmission too. |
| 00:32.53 | Chotaire | also, it's not the same if you use like a CLI test client on your asterisk box, as that would be originated from the same box. |
| 00:32.55 | *** join/#asterisk nighty^ (~nighty@210.188.173.245) |
| 00:33.27 | gregd | Chotaire: ok, when i try to conenct a sip client got this error: http://pastebin.com/ELp5mxYE |
| 00:34.00 | Chotaire | wasn't that user 6002? |
| 00:34.41 | gregd | a yeah, got two users defined.. the same way,... can recreate the same error with 6002 |
| 00:34.43 | Chotaire | SIP/2.0 404 Not found (unknown domain) |
| 00:35.29 | [TK]D-Fender | gregd: "sip show peer 6002" |
| 00:35.47 | Chotaire | and "sip show domains" |
| 00:35.58 | gregd | [TK]D-Fender: http://pastebin.com/Fjy9ayw2 |
| 00:36.24 | [TK]D-Fender | gregd: What client? |
| 00:36.33 | gregd | Chotaire: http://pastebin.com/etGWsa3L |
| 00:36.49 | Chotaire | that would be "sip show peer 6001" |
| 00:37.40 | [TK]D-Fender | yeah |
| 00:38.25 | gregd | show peer 6001: http://pastebin.com/Sq4fk508 |
| 00:38.51 | [TK]D-Fender | gregd: What client? |
| 00:39.01 | [TK]D-Fender | GrePlease show the COMPLETE attempt |
| 00:39.08 | [TK]D-Fender | gregd: Please show the COMPLETE attempt |
| 00:39.09 | gregd | [TK]D-Fender: http://code.google.com/p/telephone/ |
| 00:39.41 | [TK]D-Fender | Greyou shouldn't fill in a domain... |
| 00:39.47 | [TK]D-Fender | gregd: Just point to your * server |
| 00:40.42 | gregd | [TK]D-Fender: do you mean in my client configuration to leave domain empty? |
| 00:41.10 | [TK]D-Fender | yes |
| 00:41.44 | Chotaire | also try to comment out domain= in sip.conf |
| 00:42.00 | Chotaire | or comment... whatever... ;domain= |
| 00:42.22 | Chotaire | <- no engalisch |
| 00:42.23 | gregd | [TK]D-Fender: here is the complete request http://pastebin.com/qSuVQzMQ |
| 00:42.47 | Chotaire | SIP/2.0 404 Not found (unknown domain) |
| 00:43.25 | gregd | ok, i commented out 'domain' in sip.conf and now the client connects |
| 00:43.36 | gregd | and there are no retransmissions :) |
| 00:43.46 | gregd | why were i supposed to comment out the domain in 'sip.conf' ? |
| 00:43.49 | Chotaire | then place a call. |
| 00:43.59 | Chotaire | gregd: just because we were in the mood today. |
| 00:44.12 | Chotaire | autodomain is set by default. |
| 00:44.25 | gregd | autodomain got set to no |
| 00:44.33 | gregd | but the call did not went through |
| 00:44.41 | Chotaire | pastebin please... |
| 00:46.17 | gregd | http://pastebin.com/fw1bnJT9 |
| 00:46.22 | *** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 00:48.53 | Chotaire | SIP/2.0 401 Unauthorized |
| 00:49.24 | gregd | what means? |
| 00:49.39 | [TK]D-Fender | Chotaire: now your auth is just wrong |
| 00:50.14 | gregd | but in line 106 it is registered.. |
| 00:50.26 | gregd | it might be there the top lines are not relevant |
| 00:50.29 | Chotaire | fender: not mine ;) but wait... i was too quick.. still reading. |
| 00:51.28 | [TK]D-Fender | I see it OK'ing a bunch and refusing a bunch |
| 00:51.43 | [TK]D-Fender | 109 = success |
| 00:52.13 | [TK]D-Fender | And we see the call being accepted |
| 00:53.56 | Chotaire | it's all fine until it starts retransmitting after ringing... |
| 00:54.31 | *** join/#asterisk Arsenick- (~y@modemcable022.82-21-96.mc.videotron.ca) |
| 00:57.22 | gregd | so... ? |
| 00:58.24 | gregd | but now I've connected one more client to my asterisk box and I can ring each other perfectly fine |
| 00:58.27 | Chotaire | in sip.conf, what did you configure in fromdomain for darmowytelefon-out |
| 00:58.51 | *** join/#asterisk pabelanger (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 00:58.54 | gregd | Chotaire: it's not set |
| 00:59.09 | gregd | what should it be set to? |
| 00:59.18 | Chotaire | try fromdomain=stun.darmowytelefon.pl |
| 01:00.07 | gregd | i dont think they have stun. |
| 01:00.15 | gregd | but anyway i tried and it does not work |
| 01:00.17 | *** join/#asterisk hipitihop (~denis@203.132.229.236) |
| 01:00.17 | Chotaire | try do |
| 01:00.20 | Chotaire | oops |
| 01:00.22 | Chotaire | they do.. |
| 01:00.44 | Chotaire | try sip.darmowytelefon.pl then |
| 01:00.53 | Chotaire | that's both different addresses. |
| 01:01.53 | gregd | no, no result |
| 01:02.40 | gregd | but got again retransmits |
| 01:02.45 | *** join/#asterisk boodu (~antoine@175.158.129.128) |
| 01:02.49 | boodu | hello |
| 01:02.52 | gregd | http://pastebin.com/epzYyuP8 |
| 01:03.05 | Chotaire | lol: http://pastebin.com/WDJ9jPeh |
| 01:04.07 | *** join/#asterisk mindCrime (~chatzilla@64.134.184.64) |
| 01:04.09 | [TK]D-Fender | gregd: What do you have forwarded to you PBX? |
| 01:04.25 | gregd | my PBX is on DMZ |
| 01:04.41 | [TK]D-Fender | gregd: What router? |
| 01:04.59 | gregd | [TK]D-Fender: some dlink running dd-wrt |
| 01:05.02 | [TK]D-Fender | EW |
| 01:05.06 | [TK]D-Fender | Prime offended |
| 01:05.12 | [TK]D-Fender | DDWRT AND the D-Link |
| 01:05.24 | [TK]D-Fender | Horrible NAT implementation. screws with ports |
| 01:05.32 | [TK]D-Fender | replace and retest |
| 01:05.59 | Chotaire | I thought you had connected the asterisk box DIRECTLY to dsl? |
| 01:06.06 | Chotaire | DMZ = NO DIRECT INTERNET CONNECTION |
| 01:06.26 | gregd | lol |
| 01:06.34 | gregd | right back... |
| 01:07.03 | ruben23 | hi guys whats the problem when call is stablished but the caller are not barely heared by the called party..? |
| 01:07.06 | *** join/#asterisk coppice (~chatzilla@19.176.64.202.dyn.pacific.net.hk) |
| 01:07.33 | *** join/#asterisk gregd (~gregd@cpc9-sgyl29-2-0-cust103.sgyl.cable.virginmedia.com) |
| 01:07.49 | Chotaire | if the caller can be heard barely (but can be heard), then I'd say the telephone receiver is upside down. |
| 01:09.40 | Chotaire | gregd: use pppoe passthrough and then whatever pppoe/pptp daemon to authenticate to dsl and receive public IP directly on the ethernet interface.. or use different router, always making sure you are forwarding ALL ports (for testing) to the asterisk box. |
| 01:09.59 | grungies1138 | http://pastebin.com/YetPQn0g - no errors and no sound |
| 01:10.04 | *** join/#asterisk gregd (~gregd@cpc9-sgyl29-2-0-cust103.sgyl.cable.virginmedia.com) |
| 01:10.35 | ruben23 | Chotaire: what you mean..? |
| 01:10.52 | *** join/#asterisk voxter (~voxter@76.77.73.130) |
| 01:11.35 | Chotaire | ruben23: what protocol? |
| 01:12.08 | gregd | [TK]D-Fender: should I believe you? ;) |
| 01:12.28 | Chotaire | greg: ? |
| 01:12.57 | Chotaire | gregd: use pppoe passthrough and then whatever pppoe/pptp daemon to authenticate to dsl and receive public IP directly on the ethernet interface.. or use different router, always making sure you are forwarding ALL ports (for testing) to the asterisk box. |
| 01:13.13 | ruben23 | Chotaire: SIP...g729 codec |
| 01:13.33 | Chotaire | ruben23: which telephone/client? |
| 01:13.38 | *** join/#asterisk smooth_penguin (~smoove@115.118.216.1) |
| 01:16.14 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 01:18.26 | gregd | Chotaire: ok, how to delete such iptable rule in INPUT chain: DROP udp -- anywhere anywhere udp dpt:route ? |
| 01:19.04 | ruben23 | Chotaire: calling using softphones-caling to US |
| 01:19.07 | ruben23 | form asia |
| 01:19.22 | ruben23 | using voip trunk..also |
| 01:19.34 | *** join/#asterisk diegomad (~mad@190.158.77.2) |
| 01:19.39 | Chotaire | ruben23: do you also have low volume if you use your softphone not with your asterisk server but when registering to a voip provider? |
| 01:20.09 | *** join/#asterisk gregd (~gregd@cpc9-sgyl29-2-0-cust103.sgyl.cable.virginmedia.com) |
| 01:20.22 | ruben23 | Chotaire: havent tested it yet.. |
| 01:20.32 | grungies1138 | anybody see my pastebin? |
| 01:20.46 | ruben23 | but the asterisk server is doing predictive dialing, call volumes are a bit high. |
| 01:21.48 | Chotaire | ruben23: test it, because then it's a definitive issue with your softphone or microphone. there is sip providers who offer test calls. |
| 01:22.40 | *** join/#asterisk gregd (~gregd@cpc9-sgyl29-2-0-cust103.sgyl.cable.virginmedia.com) |
| 01:23.10 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 01:24.23 | Lantizia | I'm behind a NAT... my asterisk system is behind another NAT. Two way audio works fine. However I have nothing in sip_nat.conf... W_T_F IS GOING ON |
| 01:24.34 | Lantizia | I know it's an odd way to phrase a question :) |
| 01:24.43 | Chotaire | gregd: iptables -L -n -v, then paste the rule again. |
| 01:24.48 | Lantizia | I thought that was impossible |
| 01:25.29 | Chotaire | lantizia: then both endpoint routers are clever enough and do great connection tracking. |
| 01:25.43 | Lantizia | Chotaire, think my NAT router is rewriting the sip headers? |
| 01:26.03 | grungies1138 | I'm using mpg123 to play MOH files and I get no errors, but I have no audio. |
| 01:26.04 | gregd | ok, thanks for your help.. i will try to rearrange my routers tomorrow and try it again |
| 01:27.07 | Chotaire | lantizia: I don't know what routers you use or if you use stun servers or whatever proxies/firewalls inbetween. |
| 01:27.15 | Lantizia | no stun server |
| 01:27.23 | Lantizia | the router is Untangle |
| 01:27.34 | Chotaire | gregd: that firewall rule is dropping udp "route" packets |
| 01:27.46 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 01:27.48 | Lantizia | I am impressed :) |
| 01:29.07 | [TK]D-Fender | sip_nat.conf <- this is not inherently an * config file |
| 01:29.23 | [TK]D-Fender | [21:25]<Lantizia>Chotaire, think my NAT router is rewriting the sip headers? <- Possible |
| 01:29.39 | [TK]D-Fender | Lantizia: Nowhere do I see you actually looking at what is happening. |
| 01:29.42 | Chotaire | lantizia: untangle can do conntrack and connmark. it is possible. |
| 01:31.06 | Lantizia | If I disable SIP Helper in Untangle then I only get 1 way audio :) |
| 01:31.16 | Lantizia | Nifty :) |
| 01:31.28 | Chotaire | SIP Helper.. says it all. |
| 01:31.46 | Chotaire | Seems a SIP Connection Tracker/Proxy Firewall Module or similar. |
| 01:32.11 | Lantizia | Co co co co co co co coooool |
| 01:32.29 | Chotaire | indeed. |
| 01:32.32 | *** join/#asterisk slvrchair (~Slvrchair@ool-18e410d7.dyn.optonline.net) |
| 01:32.37 | *** join/#asterisk ming_zym (~ming_zym@114.251.86.0) |
| 01:32.58 | *** join/#asterisk sourcode (~code@ppp-58-8-110-110.revip2.asianet.co.th) |
| 01:33.27 | slvrchair | Hoping someone can help me out... I upgraded asterisk using "yum update" and now none of my phones or trunks will register. Any troubleshooting suggestions? Asterisk seems to be running and everything. |
| 01:35.10 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 01:36.39 | Chotaire | Lantizia: I'm a networker. I'll check it out these days, just grabbed it. |
| 01:37.03 | Lantizia | It's very nice :) I suppose I should have guessed earlier it was that |
| 01:37.19 | Chotaire | lantizia: all these (incredibly expensive) pay features you can have for free tho |
| 01:37.24 | Lantizia | I used it virtualised and it protects my virtual machines |
| 01:38.02 | Lantizia | The phone system went from a non-NAT physcial server to a NAT virtual server about a month ago and I only just realised that it just worked without reconfiguration! lol |
| 01:39.09 | Chotaire | you only just realized? nobody even checked if the phone system was working for a whole month? ;) |
| 01:39.46 | Lantizia | oh sure... but it only just occurred to me... WTF how is it doing this!? MY BRAIN IS MELTING |
| 01:40.00 | Lantizia | it's 2:40am and I'm tired lol |
| 01:40.40 | *** part/#asterisk AI_Rob (~rob@c-76-127-249-66.hsd1.ct.comcast.net) |
| 01:41.04 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 01:41.17 | boodu | re |
| 01:41.29 | boodu | I try to connect a cisco phone to elastix |
| 01:42.11 | boodu | it's 7906 with the firmware SIP11.8-3-1S |
| 01:42.41 | boodu | i configure my SEPXXX.cnf.xml file |
| 01:42.59 | boodu | i have tonality |
| 01:43.20 | boodu | but the phone is not connected on my asterisk |
| 01:45.55 | Lantizia | nighty night |
| 01:45.59 | Lantizia | nice chatting to you Chotaire :) |
| 01:47.09 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 01:47.47 | boodu | http://pastebin.org/300217 <-- my SEPXXX.cnf.xml |
| 01:49.47 | boodu | when there are not proxy adress [ <proxy>192.168.1.40</proxy> ] the phone don't detect the Pabx |
| 01:49.53 | boodu | and here it's curious |
| 01:50.29 | boodu | the auth login and password are wrong but the phone don't detect it |
| 01:51.14 | boodu | I need of your help please |
| 01:52.21 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 01:52.53 | slvrchair | I upgraded asterisk and am getting this error... any ideas? chan_sip.c: Unable to load config sip.conf |
| 01:53.02 | *** join/#asterisk trnzmeta (~bleh@secure27.lnk.telstra.net) |
| 01:57.39 | *** join/#asterisk pabelanger (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 02:01.02 | *** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net) |
| 02:01.03 | vader-- | hello |
| 02:03.44 | ChannelZ | may the force be with you |
| 02:04.39 | [TK]D-Fender | WHO'S YOUR DADDY?! |
| 02:05.51 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 02:05.52 | vader-- | is the SMDI module something i need? |
| 02:06.00 | vader-- | im not interfacing with other PBX systems |
| 02:08.22 | ChannelZ | no though if I remember correctly there is some sort of internal bit of smdi that is used for something else... |
| 02:08.59 | ChannelZ | voicemail or maybe MWI, I'm drawing a blank |
| 02:11.00 | *** join/#asterisk coppice (~chatzilla@m121-202-75-229.smartone-vodafone.com) |
| 02:13.10 | *** join/#asterisk swathanthran (~user@unaffiliated/shyam-k/x-8459115) |
| 02:15.07 | swathanthran | internal modems, used to get dialup connections can work with asterisk equivalent to a x100p ? |
| 02:15.29 | swathanthran | like say i am having a softmodem, one which is embedded in soundcard or so. |
| 02:15.51 | ChannelZ | no |
| 02:16.03 | swathanthran | they are just different hardwares right? |
| 02:16.49 | ChannelZ | yes |
| 02:18.02 | swathanthran | i was doubting if its that the hardware is doing something else or lack of drivers for those? |
| 02:18.35 | ChannelZ | Probably a bit of both but mostly the latter |
| 02:18.36 | coppice | its a lack of drivers. want to write some? |
| 02:21.27 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-54.mia.bellsouth.net) |
| 02:22.00 | swathanthran | well.. i don't know anything as of now, have been pushing to do something interesting, and this would be pretty greatly interesting if its attainable to do that for softmodems, as i guess they will still be in production as its nicely integrated with sound cards.. |
| 02:25.15 | grungies1138 | I'm having issues with my moh |
| 02:25.37 | grungies1138 | I took 3 mp3s and converted them to u-law wavs |
| 02:25.37 | coppice | there is some form of linux software for many of the common softmodems - binary DSP code and source for the actual hardware driver. that source code is the basis you need for a driver to integrate with dadhi. its not that tough or that big a jon, yet after 10 years of *, nobody has done it |
| 02:26.03 | swathanthran | coppice: any one tried that already or some pages to get more info on asterisk+softmodems? |
| 02:26.42 | [TK]D-Fender | ZAPRAS <- |
| 02:26.51 | swathanthran | oh ic ah to integrate with dahdi.. great.. i'll surely read on tha.. |
| 02:28.51 | grungies1138 | <PROTECTED> |
| 02:31.01 | *** join/#asterisk mhaddog (~MhaddogM1@z65-50-118-232.ips.direcpath.com) |
| 02:32.19 | *** part/#asterisk mhaddog (~MhaddogM1@z65-50-118-232.ips.direcpath.com) |
| 02:32.28 | TJNII | permissions. |
| 02:32.34 | grungies1138 | ahh! |
| 02:34.51 | grungies1138 | nope |
| 02:38.15 | grungies1138 | hmm. different error now. what format of wav does it expect? |
| 02:42.07 | grungies1138 | unable to open format |
| 02:43.49 | *** join/#asterisk maldous (~maldous@scooby.unicity.com.au) |
| 02:43.57 | maldous | 'lo |
| 02:48.44 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:54.21 | maldous | question here about http://pastebin.com/KgAvnk8v |
| 02:55.15 | maldous | i have asterisk registered as a sip client, want received calls to dial another sip client, but i'm having no luck. any suggestions? |
| 03:03.36 | mmlj4 | which sip proxy is sane these days? ser, openser, opensips? |
| 03:11.10 | [TK]D-Fender | maldous: You seem to have forgotten the concept of CONTEXT's and you need to pay attention to what the CALLER is dialing in. |
| 03:11.22 | [TK]D-Fender | maldous: enable SIP DEBUG and watch the INVITE that comes in. |
| 03:16.00 | [TK]D-Fender | maldous: exten => AAAA,1,Dial(SIP/FFFF@BBBB) <--- and since this is so vauge... who knows what wrong in the real version... |
| 03:16.05 | [TK]D-Fender | vague* |
| 03:18.10 | slvrchair | Does anyone have any idea what could be causing this error? chan_sip.c:24274 reload_config: Unable to load config sip.conf |
| 03:24.23 | *** join/#asterisk coppice (~chatzilla@m121-202-75-229.smartone-vodafone.com) |
| 03:24.37 | ChannelZ | uh yeah.. the inability to read or possibly parse your sip.conf |
| 03:24.52 | ChannelZ | It's a completely esoteric error message, I know.. |
| 03:25.27 | slvrchair | I did "yum update" and it updated a bunch of asterisk stuff... I'm also using FreePBX... so I'm guessing the new asterisk doesn't like my sip.conf |
| 03:25.42 | slvrchair | Although I also did "yum install asterisk16-configs" and it put a whole new set of config files in, and it still doesn't like those |
| 03:25.57 | slvrchair | Is there a way to get some more useful error messages on what is actually failing? |
| 03:27.39 | [TK]D-Fender | slvrchair: And you should NOT have put in new configs. |
| 03:27.52 | [TK]D-Fender | slythose will annihilate FreePBX's work. |
| 03:28.03 | [TK]D-Fender | slvrchair: And potentially cripple it from rebuilding |
| 03:29.32 | slvrchair | i still have all my old configs so i can stick them back in, or have freepbx do it |
| 03:30.25 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 03:32.37 | slvrchair | i think the problem is, "yum update" installed freepbx 2.7.0.2, and then i ran the "module admin" update in freepbx and it downgraded it to 2.7.0.0... |
| 03:33.42 | slvrchair | After the "yum update" the FreePBX Framework went from "Enabled" to "Disabled; Pending upgrade to 2.7.0.0" |
| 03:33.46 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 03:36.25 | [TK]D-Fender | slvrchair: Put your configs back. Verify the permissions, and run "amportal chown" |
| 03:36.35 | [TK]D-Fender | slvrchair: And then move over to #freepbx |
| 03:36.36 | [TK]D-Fender | ~freepbx |
| 03:36.37 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 03:36.39 | [TK]D-Fender | ^^^^^^^^ |
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| 03:43.09 | slvrchair | D-Fender: Will do, thanks |
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| 04:01.00 | grungies1138 | anybody willing ot help me with Music on hold issues? |
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| 04:45.31 | darkdrgn2k | hi all. Is there a software to test connection quality (ie jitter and such) of a connection? |
| 04:52.47 | *** join/#asterisk Keal (~jargon@unaffiliated/jargon) |
| 04:53.01 | Keal | Does Asterisk come with .py scripting support? |
| 04:53.39 | russellb | Keal: not natively, but look up the AGI interface |
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| 04:54.26 | devdvd | hey all, having a bit of an issue. trying to store my agents in mysql but for some reason the ackcall function isn't working. It works fine when the agents are staticaly defined (no realtime) but not when pulled from the database |
| 04:54.28 | devdvd | any thoughts? |
| 04:54.44 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:54.55 | [TK]D-Fender | devdvd: Yeah.. I think you should SHOW US |
| 04:55.20 | [TK]D-Fender | russellb: that may as well be a "yes" :) |
| 04:55.31 | [TK]D-Fender | russellb: Sure he can! |
| 04:55.43 | devdvd | when the agents are static (in agents.conf/queues.conf) i see this in the console: |
| 04:55.44 | devdvd | <PROTECTED> |
| 04:56.10 | devdvd | but when its pulled from mysql, I dont see that |
| 04:56.12 | [TK]D-Fender | devdvd: pastebin a COMPLETE call. |
| 04:56.14 | devdvd | and the call just connects |
| 04:56.15 | devdvd | sure |
| 04:56.18 | [TK]D-Fender | devand configs |
| 04:58.02 | russellb | [TK]D-Fender: *nods* ... when i hear that, though, i think of embedding the python interpreter, which we don't do |
| 04:58.20 | russellb | though I have for fun :-) |
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| 05:00.53 | Keal | russellb, what is the point of Asterisk if it can't be used with Python? |
| 05:01.09 | [TK]D-Fender | russellb: Round hole, square peg.... meet russellb+big hammer :p |
| 05:02.45 | Keal | perl is nearing obsolete. |
| 05:03.02 | russellb | Keal: heh, but you can ... |
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| 05:03.26 | Keal | How do I fit Asterisk with Python support, then? |
| 05:03.39 | russellb | did you look up AGI? |
| 05:03.59 | Keal | This is for a Windows 7 Pro / quad 2.13GHz/3.6GHz Core i3 330M machine |
| 05:04.04 | russellb | are you serious? |
| 05:04.10 | Keal | Yes. |
| 05:04.16 | [TK]D-Fender | \o/ |
| 05:04.21 | [TK]D-Fender | Keal: meaningless |
| 05:04.31 | Keal | Is there either amd64 or ia64 build of Asterisk? |
| 05:04.33 | russellb | asterisk is not supported on windows. |
| 05:04.40 | Keal | This laptop supports both architectures. |
| 05:05.58 | Keal | It had better be, or a good portion of the world would have no use for it. |
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| 05:06.28 | russellb | Keal: sorry. |
| 05:07.09 | Keal | Windows is more stable than Linux now. |
| 05:07.16 | russellb | k |
| 05:07.43 | [TK]D-Fender | Keal: What good portion of the world needs a PBX & telephony toolkit to run on their Windows 7 laptop? |
| 05:08.06 | Keal | Anyone that doesn't live anywhere. |
| 05:08.09 | [TK]D-Fender | Keal: Oh... Running UNDER Windows. Not that you have to anyway. |
| 05:09.00 | devdvd | ~pastebin |
| 05:09.00 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 05:10.03 | russellb | the flood of explaining pastebin is almost as annoying as an actual paste |
| 05:12.53 | devdvd | well, i didnt really need pastebin explained...i just needed to know what one was accepted |
| 05:13.19 | devdvd | some support channels go completly freaking scitzo if you use the wrong one |
| 05:13.22 | devdvd | http://pastebin.com/fnchYMWx |
| 05:13.26 | devdvd | theres the pastebin |
| 05:13.54 | russellb | ha, it's all good |
| 05:14.59 | devdvd | TK, i included console output from 2 calls, one with a static agent and one with a realtime agent, i also included extensions.conf, queues.conf, agents.conf, and extconfig.conf |
| 05:16.21 | devdvd | please excuse the funny looking dialplan. I tend to make things work the best way i know how then learn better ways later |
| 05:21.23 | [TK]D-Fender | devdvd: How do you have your agents when not in realtime? |
| 05:21.30 | [TK]D-Fender | devI don't see that part |
| 05:24.11 | devdvd | oh, sorry, just a sec |
| 05:24.14 | [TK]D-Fender | devdvd: that aside the rest does seem to match up 100%. What ver of *? |
| 05:24.30 | devdvd | 1.4.31\ |
| 05:24.38 | [TK]D-Fender | :/ |
| 05:24.59 | devdvd | im working on the upgrade to 1.6 :) |
| 05:25.06 | [TK]D-Fender | devdvd: Ok, perhaps something else will stand out for others, but its bed-time for Bonzo here... |
| 05:25.12 | [TK]D-Fender | later all, BBT |
| 05:25.15 | devdvd | ok thanks tk |
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| 05:27.07 | russellb | devdvd: i'll look at the code real quick .. |
| 05:27.35 | kruemeltee | hello again :-) |
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| 05:29.08 | devdvd | just a sc and ill show u the diff in the configs for realtime vs static |
| 05:29.11 | devdvd | sec |
| 05:29.18 | russellb | k |
| 05:30.14 | russellb | static realtime? |
| 05:30.20 | russellb | chan_agent does not appear to have any realtime support |
| 05:30.28 | russellb | the dynamic realtime stuff i mean |
| 05:31.36 | devdvd | hmm..interesting |
| 05:31.44 | devdvd | cuz i can define an agent in realtime |
| 05:31.56 | devdvd | and * will connect with it out of the queue |
| 05:31.59 | russellb | you mean defining queue members in realtime? |
| 05:32.05 | devdvd | yes |
| 05:32.14 | russellb | yeah, that's different ... defining agents, and then putting agents in queues |
| 05:32.35 | devdvd | well, |
| 05:32.41 | devdvd | | uniqueid | membername | queue_name | interface | penalty | paused | |
| 05:32.42 | devdvd | +----------+------------+------------+------------------------------------+---------+--------+ |
| 05:32.42 | devdvd | | 1 | jason | support | Local/217127300881@outbound2agents | 1 | 0 | |
| 05:32.46 | devdvd | is my db entry |
| 05:33.00 | devdvd | and i dont supply that anywhere in agents.conf or queues.conf |
| 05:33.37 | devdvd | and * will connect without issue |
| 05:33.56 | devdvd | maybe im misunderstanding what you are saying? |
| 05:33.57 | russellb | ah, I see what you've got going on |
| 05:34.04 | devdvd | :) |
| 05:34.06 | devdvd | cool |
| 05:34.08 | devdvd | i think |
| 05:34.17 | russellb | in the first example output, app_queue is calling Agent/869 |
| 05:34.30 | russellb | in the second example (where you're using realtime), you're not using chan_agent in there at all |
| 05:34.37 | russellb | the ackcall feature is a chan_agent thing |
| 05:34.51 | devdvd | ahhh |
| 05:35.08 | devdvd | and chan_agent has no rt support so what im trying to do wont work/ |
| 05:35.26 | russellb | well, you can still do it, but in your table where you have Local/..., you need Agent/... |
| 05:35.34 | russellb | they're not equivalent setups |
| 05:35.49 | russellb | what you have told app_queue to dial to reach an agent is different in these 2 cases |
| 05:37.05 | russellb | i never saw the config of your queue members without realtime |
| 05:37.23 | russellb | but I presume you told it Agent/869 is a queue member, and not Local/217...... |
| 05:37.32 | devdvd | yes |
| 05:37.38 | devdvd | Agent/869 in queuesl.conf |
| 05:37.46 | russellb | so, see what i'm saying? |
| 05:37.48 | devdvd | yea |
| 05:37.52 | russellb | k |
| 05:38.34 | devdvd | thats gonna break a major thing though and pretty much defeat the purpose of using realtime. so i have to decide rather the benefits of realtime outweigh not having that ack |
| 05:39.13 | devdvd | does chan_agent have rt support in 1.6? |
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| 05:39.27 | russellb | no |
| 05:39.32 | devdvd | ok |
| 05:39.34 | russellb | but you might be able to emulate the ackcall feature in the dialplan |
| 05:39.48 | devdvd | ya, im thinking about that very thing |
| 05:39.55 | russellb | using the M() option to Dial() |
| 05:40.04 | russellb | execute a macro that asks if they want to accept |
| 05:40.15 | devdvd | hmm |
| 05:40.20 | devdvd | that might just work |
| 05:40.32 | russellb | and if they don't accept, set the MACRO_RESULT variable to BUSY |
| 05:40.36 | devdvd | that might just work indeed :) |
| 05:40.37 | russellb | and then it will be like they didn't answer |
| 05:40.48 | russellb | ta-da |
| 05:41.04 | russellb | and if you want to be fancy, you could use the Verbose() application in the dialplan to emulate the message you saw in your log from chan_agent, heh |
| 05:41.22 | devdvd | :) |
| 05:41.41 | devdvd | thank you very much |
| 05:41.44 | russellb | you're welcome |
| 05:42.15 | devdvd | wife is nagging me to go to bed now:( |
| 05:42.32 | russellb | you should probably listen |
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| 05:47.32 | jony123 | Is there any programs for realtime management of the dailplan and the configuration od the phones. |
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| 05:51.06 | jony123 | i heard about a mysql one somewhere but can't remember where or any links. |
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| 06:03.39 | jesselang | Hi. Is there any way to tell which end of the call leg hung up? |
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| 06:04.43 | mianos | depends on country |
| 06:04.50 | mianos | sometime polatiry |
| 06:04.51 | kaldemar | jesselang: app Dial option g |
| 06:05.03 | mianos | sometimes you neeed to monitor the hangup busy tones |
| 06:05.13 | mianos | oops |
| 06:05.15 | mianos | sorry |
| 06:05.17 | mianos | wrong answer |
| 06:05.24 | mianos | which end, yer I dunno :( |
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| 06:18.43 | jesselang | kaldemar, what about in the AMI events? |
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| 06:20.39 | kaldemar | jesselang: the hangup event has a channel identifier. it could be one way to determine it. where do you want the information? |
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| 06:26.17 | jesselang | kaldemar, usually there are two hangup events, one for each leg. So can I trust that the first hangup event is the leg that hung up the call? |
| 06:27.24 | jesselang | I'd love if the unlink event provided some definite information about which leg ended the call. |
| 06:28.36 | kaldemar | jesselang: i won't guarantee you the event order since i don't know the code, but you can always test it. |
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| 06:30.18 | jesselang | Sounds like fun for tomorrow. Thanks kaldemar! |
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| 07:31.06 | ijpalmer | hi does anyone know why MixMonitor would stop recording when I complete a transfer onan outbound call, I'm using AUDIOHOOK_INHERIT=yes. MixMonitor doesn't stop recording when inbound transferred calls are completed |
| 07:33.21 | devdvd | ijpalmer: i remember reading something about that. Has to do something with the channel being destroyed when transfered |
| 07:34.57 | ijpalmer | devdvd: thanks for your reply, yeah I found that also but then AUDIOHOOK_INHERIT should keep MixMonitor recording, which it does on inbound calls but not outbound |
| 07:36.46 | devdvd | sorry, i dont have an answer for that |
| 07:37.47 | ijpalmer | ok thanks devdvd, looks like I'm gonna have to keep looking |
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| 08:20.21 | MAbbas | Hi All, I am facing problem in attended call transfer. When I attended transfer to an extension. I get error saying "I am sorry thats not a valid extension. Please try again". http://pastebin.com/cZd3eqjx |
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| 08:46.59 | nicked | hi guys, how do i block outgoing an outgoing idd number? |
| 08:48.38 | ijpalmer | hi does anyone know why MixMonitor would stop recording when I complete a transfer on an outbound call, I'm using AUDIOHOOK_INHERIT=yes. MixMonitor doesn't stop recording when inbound transferred calls are completed |
| 08:59.58 | nalbax | hi |
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| 09:05.48 | nalbax | i have an TAPI error when i start asterisk win32, when asterisk load chan_tapi.so, i have error "win32_tapi.c:129 telephonyInitialize: No suitable line found". somebody have an idea for correct this ? |
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| 09:30.17 | Lantizia | BUMPER CARZ |
| 09:30.21 | Lantizia | Morning :) |
| 09:33.06 | Lantizia | Is there an official place I can lookup the API for the AMI? (e.g. all particular actions relevant to 1.4) |
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| 09:37.46 | gregd | Chotaire: hey, are u there? |
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| 10:16.39 | niekvlessert | hi, can i make asterisk give a sip 302 message? |
| 10:19.19 | *** join/#asterisk mallchin (~mallchin@mail.dataproservices.co.uk) |
| 10:20.01 | mallchin | hi guys -- having issues compile zaptel-1.4.11 on a recent kernel, found a patch but it still fails compiling, what to do? |
| 10:29.38 | kaldemar | niekvlessert: core show application Transfer & promiscredir parameter in sip.conf |
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| 10:34.40 | kaldemar | mallchin: zaptel has gotten a lot of fixes since 1.4.11. the newest one is 1.4.12.1. and, zaptel has been renamed to DAHDI almost 2 years ago. any particular reason for not using DAHDI? |
| 10:38.34 | mallchin | kaldemar: we use zaptel in a production environment -- what is required to switch to dhadi? |
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| 10:39.00 | mallchin | kaldemar: if it is a simple swapout without any changes to code or config files I could use dhadi instead |
| 10:39.23 | kaldemar | mallchin: a version of asterisk that supports it, i.e. 1.4.22 or newer. |
| 10:40.01 | kaldemar | mallchin: the config is pretty much the same syntax, mostly changes in file locations and names. |
| 10:40.31 | kaldemar | Zaptel-to-DAHDI.txt in the source package will tell some more. |
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| 10:41.15 | mallchin | kaldemar: that could cause us problems, but I'll give it a go if it is the last resort |
| 10:41.41 | kaldemar | try 1.4.12.1 first then. |
| 10:42.03 | mallchin | kaldemar: is there any way to get zaptel to compile? I saw a patch but it still fails compiling |
| 10:43.36 | kaldemar | feel free to pastebin an error, maybe someone will take a look. |
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| 11:14.07 | DND | guys i get this error in my logs |
| 11:14.10 | DND | mpg123: Can't rewind stream by 20 bits! |
| 11:14.16 | DND | what does that mean |
| 11:14.34 | DND | also this one: mp3/interface.c: Junk at the beginning of frame 49443303 |
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| 11:47.35 | Lantizia | Does anyone know of an AMI action other than QueuePause that FOP2 picks up and can show an indicator for? |
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| 12:20.42 | DelphiWorld | hi |
| 12:21.04 | DelphiWorld | anyone from belgium? |
| 12:23.03 | [TK]D-Fender | No, but I like their waffles :) |
| 12:24.34 | DelphiWorld | [TK]D-Fender: or mayb you can do... what my need :P |
| 12:25.18 | DelphiWorld | [TK]D-Fender: are you in US? |
| 12:27.52 | [TK]D-Fender | DelphiWorld: Canada |
| 12:29.51 | DelphiWorld | [TK]D-Fender: lol... unfortunatly... no carrier do that in canada :P |
| 12:29.55 | t_dot_zilla | screw belgium with their fruity beer |
| 12:30.13 | DelphiWorld | t_dot_zilla: lol |
| 12:30.19 | DelphiWorld | t_dot_zilla: and u where r u from? |
| 12:30.25 | t_dot_zilla | buffalo, NY |
| 12:30.46 | t_dot_zilla | i assume you are from belguim? |
| 12:31.04 | DelphiWorld | t_dot_zilla: no;) |
| 12:31.07 | DelphiWorld | t_dot_zilla: algeria |
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| 12:31.15 | t_dot_zilla | cool |
| 12:31.25 | t_dot_zilla | are you french? |
| 12:31.42 | [TK]D-Fender | DelphiWorld: Do what? |
| 12:32.13 | DelphiWorld | [TK]D-Fender: INUM |
| 12:32.29 | DelphiWorld | t_dot_zilla: no, arabic... and french... and en... |
| 12:32.49 | t_dot_zilla | half imperialist |
| 12:34.17 | DelphiWorld | i need someone that have verison or belgacom |
| 12:34.26 | *** join/#asterisk af_ (~getsmart@88-149-241-176.dynamic.ngi.it) |
| 12:35.05 | DelphiWorld | or evean BT |
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| 12:47.11 | suma | in a dial command, on connection we can get it to a Macro, if two channels refer to the same variable, will it be overwritten ? |
| 12:47.42 | suma | VAR1 I'm using it for the first channel, the same variable is used for the second channel |
| 12:48.04 | suma | I mean the variable name |
| 12:48.31 | suma | If i use VAR1 which context it will refer to, will it be to first channel or the second channel ? |
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| 12:50.03 | devdvd | suma: i am not 100% sure but id assume as with the law of variables, the last channnel to set it is the value it takes |
| 12:50.29 | devdvd | now |
| 12:50.45 | devdvd | that said, are these gonna be per-call variables? |
| 12:51.23 | suma | when i refer a variable it also refers to which channel it belongs to right ? |
| 12:51.35 | devdvd | like say you have s,1,Answer() s,n,Set(blah=1) and that variable will persist throughout that call |
| 12:51.44 | devdvd | if so then i dont believe it will be overwriten by another call |
| 12:52.01 | devdvd | otherwise the use of variables would be silly |
| 12:52.42 | devdvd | actually come to think of it (which im not doing very well of atm cuz ive had like no sleep) |
| 12:53.22 | devdvd | all variables are per-call variables |
| 12:53.43 | suma | let me check the source, please one moment |
| 12:55.17 | suma | pbx_bulitin_getvar_helper needs channel to retrieve a variable |
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| 12:57.01 | suma | i don't know how it is manipulated per call |
| 12:57.30 | [TK]D-Fender | [08:49]<devdvd>suma: i am not 100% sure but id assume as with the law of variables, the last channnel to set it is the value it takes <- no |
| 12:57.49 | [TK]D-Fender | these are CHANNEL VARIABLE. They exist within the scope and uration of the CHANNEL |
| 12:57.56 | [TK]D-Fender | +d |
| 12:58.06 | suma | i'm clear now |
| 12:58.37 | suma | when it goes to macro on call connect, if I refer a variable that exists in both the channels, which variable value I will get ? |
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| 13:01.54 | [TK]D-Fender | suma: go read the channelvariables doc to understand how inheretence works |
| 13:02.03 | [TK]D-Fender | suma its a 1-way relationship. |
| 13:03.55 | suma | [TK]D-Fender: I will read the doc now, thanks for the pointer. I'm getting the results of the call in the 200 OK sip header and setting them in a variable. Trying to retrieve them in the macro. |
| 13:05.59 | suma | [TK]D-Fender: Inheritance is fine, I would it refers to the first channel |
| 13:06.07 | suma | How would I get the value of the second channel ? |
| 13:06.23 | suma | * I would see it refers to the first channel |
| 13:06.56 | suma | I mean the connected channel |
| 13:07.03 | [TK]D-Fender | suma: first what is this channel calling? |
| 13:07.25 | *** part/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 13:07.34 | [TK]D-Fender | suma: Also, * can't grab headers except from the initial INVITE |
| 13:08.14 | suma | [TK]D-Fender: I wrote the code to grab the value from headers and assingn it to that channel. |
| 13:08.53 | [TK]D-Fender | sumamodded chan_sip? |
| 13:08.55 | suma | When you Dial and if the call is connected, there will be two channels. Calling channel and the Called Channel |
| 13:09.06 | suma | [TK]D-Fender: yes |
| 13:09.47 | [TK]D-Fender | suma: If you did an insertion that way you would call a script that would use AMI to track the matching channel and retreive it via AMI as well, copying it back |
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| 13:11.17 | [TK]D-Fender | AT&T announced some changes to the iPhone data plans, but iPad 3G owners are affected, too: AT&T is doing away with the $30/month unlimited data plan. Instead you'll pay $25/month for 2GB <---- hahaha, already fucked |
| 13:11.41 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 13:12.06 | [TK]D-Fender | AT&T iPhone Tethering: An Extra $20/Month—If You Abandon Unlimited Data. We knew that we'd get iPhone tethering on AT&T soon and now we finally know that it'll cost us an extra $20 per month. But there's a catch: We'll have to abandon our unlimited data plans for it <- More AT&T + Apple hilarity |
| 13:12.12 | suma | Got you, that helps. I will write a custom application to get based on the channel. |
| 13:13.57 | [TK]D-Fender | suma: I would suggest you be careful about how you implement this not to get completely tied to 1 * branch.... |
| 13:15.42 | beefpastry | Verizon's "unlimited" is 5GB, tethering available on Android. |
| 13:17.08 | beefpastry | They all have their "gotchas" |
| 13:17.42 | Kobaz | if you don't buy tethering, unlimited is actually unlimited |
| 13:18.03 | Kobaz | according to many posts on forums, who have used more than 5gb and not gotten charged |
| 13:18.48 | beefpastry | Yeah...heard the same. Haven't been called in by the boss yet that I've used too much, but then again, I'm mostly on local wifi. |
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| 13:23.13 | [TK]D-Fender | beefpastry: Yes, but they didn't take your unlimited away from you, drop to 2GB and charge you MORE :p |
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| 13:29.06 | beefpastry | True...that is much worse than advertising an unlimited plan and using small print to get out of it. |
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| 13:31.03 | devdvd | Hey TK, russellb figured out why ackcall wouldn't work last night. It is because chan_agent doesn't support realtime and ackcall is a function of chan_agent |
| 13:32.23 | OlafsenM | guys, i have an issue |
| 13:32.44 | OlafsenM | 2 days ago I had some DAHDI calls hanging |
| 13:33.27 | OlafsenM | the only log i see is "Accepting call", and after couple of seconds "hangup ACK", but Asterisk didn't do anything on ACK message |
| 13:33.40 | OlafsenM | the call stayed active |
| 13:33.56 | t_dot_zilla | anybody familiar with scrolling marquee on an LED? |
| 13:34.00 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-bcbthgowbkgnjbab) |
| 13:34.01 | OlafsenM | in what case does telco send 'hangup ACK' message? |
| 13:34.19 | t_dot_zilla | like displaying queue stats in a scrolling marguee LED? |
| 13:35.45 | [TK]D-Fender | devdvd: Makes little sense. |
| 13:35.46 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:36.00 | [TK]D-Fender | t_dot_zilla: Plenty of LED panels out there. |
| 13:36.09 | [TK]D-Fender | t_dot_zilla: the * part is rather minor |
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| 13:36.34 | t_dot_zilla | how would the LED capture live data from * ? |
| 13:36.44 | [TK]D-Fender | t_dot_zilla: I get live stats on 2 queues, 4 agents, 4 phones, and 2 VM boxes directly on my phones |
| 13:36.59 | t_dot_zilla | using RSS ? |
| 13:37.00 | [TK]D-Fender | t_dot_zilla: LED doesn't "capture" anything. YOU send it what to display |
| 13:37.16 | [TK]D-Fender | t_dot_zilla: Most are serial-type devices |
| 13:37.32 | [TK]D-Fender | t_dot_zilla: And have a small micro-controller |
| 13:37.59 | t_dot_zilla | right, so you'd have create a program to send the data? |
| 13:38.46 | t_dot_zilla | have you heard of realterm? |
| 13:39.23 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:39.24 | [TK]D-Fender | t_dot_zilla: Yes, you need to program, no I don't recognize that program. |
| 13:41.19 | OlafsenM | so.. no help? |
| 13:41.34 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 13:41.48 | [TK]D-Fender | OlafsenM: My guess... after you send a hangup request. |
| 13:41.56 | OlafsenM | y, but i didnt send it |
| 13:42.09 | OlafsenM | that's what is so f****d up |
| 13:42.23 | OlafsenM | and my telco has "congestion" in his cdr for that call |
| 13:42.34 | OlafsenM | i mean, wtf |
| 13:43.01 | [TK]D-Fender | OlafsenM: Perhaps you should... oh I dunno... maybe... like kinda LOOK AT THE CALL or something. |
| 13:43.13 | OlafsenM | look at the call? |
| 13:43.26 | OlafsenM | what does that mean? |
| 13:44.04 | [TK]D-Fender | OlafsenM: You wonder about their responses... well look at your CALL |
| 13:44.24 | OlafsenM | well, i looked at my log |
| 13:44.30 | OlafsenM | and no hangup request sent |
| 13:46.04 | [TK]D-Fender | OlafsenM: logs = crap. Look live |
| 13:46.20 | OlafsenM | look live? |
| 13:46.31 | OlafsenM | this was a one-time issue, so i cant look live |
| 13:48.44 | [TK]D-Fender | OlafsenM: Oh well... no usable data, one time issue. Nothing to see, nothing to do then. |
| 13:49.02 | [TK]D-Fender | OlafsenM: load module res_fluxcapacitor.so |
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| 14:11.21 | drfreeze | Hey |
| 14:11.30 | drfreeze | Got a wierd problem that started yesterday |
| 14:12.37 | chuckf | don't worry, it only lasts a couple of days and she's back to her happy self again.... |
| 14:13.04 | drfreeze | Seems that transfers stopped working. Also, when testing this morning, we think we now understand the dropped calls. It seems that sometimes transfers are killing asterisk |
| 14:13.52 | drfreeze | this system has been in place for 2 years and hasn't been modified recently |
| 14:14.22 | *** join/#asterisk Dovid (~annon@213.8.118.62) |
| 14:14.46 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 14:17.27 | [TK]D-Fender | drfreeze: 2 year old * version? Please place the barrel firmly to your temple before pulling the trigger :p |
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| 14:17.52 | drfreeze | hey, why should that matter? |
| 14:20.10 | [TK]D-Fender | drfreeze: Because of bugs fixed in the past 2 years |
| 14:21.22 | Baylink | Well, I'm pretty sure the thrust of drfreeze's inquiry is: "why would 2 year old bugs *suddenly* cause things that worked to stop working?", but... |
| 14:21.54 | drfreeze | I don't think it is latent bugs |
| 14:22.04 | drfreeze | may be hardware TDM related |
| 14:22.17 | Baylink | drfreeze: which Asterisk version? |
| 14:22.38 | Baylink | And what interface? PRI/Zaptel? |
| 14:22.43 | drfreeze | system con1.4.2 |
| 14:22.52 | drfreeze | Asterisk 1.4.2 |
| 14:23.37 | drmessano | omg |
| 14:24.19 | drmessano | Did you restart asterisk? |
| 14:24.35 | drfreeze | Restarted the machine and asterisk |
| 14:24.52 | drmessano | Maybe it's a Y2K bug. How old is 1.4.2? |
| 14:26.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:26.32 | kaldemar | 1.4.2: 2007-03-19 |
| 14:26.47 | Baylink | Digium has posted "earliest version with no y2k bugs" lists; 1.4.2 well posted dated y2k though, and should be clean. |
| 14:27.05 | Baylink | drfreeze: what interface, and has your carrier changed anything? |
| 14:28.20 | drfreeze | Baylink: No known changes by carrier. What do you mean by interface? |
| 14:28.32 | Baylink | Analog? T-1? PRI? |
| 14:28.37 | Baylink | What card, which drivers? |
| 14:28.43 | drfreeze | Analog TDM cards |
| 14:28.49 | drfreeze | zapata drivers |
| 14:29.13 | Baylink | If they're analog on the PSTN interface, no need to call them TDM. Hmmm. |
| 14:29.35 | drfreeze | Plan on implementing PRI next month with upgraded asterisk. But no changes as of yet |
| 14:29.38 | Baylink | "Transfers are killing asterisk". Can you expand on that? |
| 14:29.38 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.62.91) |
| 14:30.27 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 14:30.28 | drfreeze | They told me transfers were not working. I had them call me. |
| 14:31.05 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-74-70.vif.net) |
| 14:31.23 | drfreeze | First tranfer attempt worked, but it was delayed by several seconds and they had to hit answer (polycom phones) to get the line active instead of just pick up the receiver as normal |
| 14:32.01 | Baylink | This is raw asterisk, no layers on top? |
| 14:32.19 | drfreeze | When trying to transfer me back to the original line, call was dropped. I was watching the console and I got kicked out. I restarted the console and saw that the uptime was 5 seconds, so it appears that the transfer back killed asterisk |
| 14:32.33 | drfreeze | right, raw asterisk with extensions.conf |
| 14:34.27 | *** join/#asterisk OlafsenM (~mark.olaf@193.198.31.85) |
| 14:34.30 | OlafsenM | hi guys |
| 14:34.39 | OlafsenM | new info on my hanging calls |
| 14:35.17 | OlafsenM | ISDN End-Of-Selection info is "52", Congestion without differentiation |
| 14:35.33 | OlafsenM | so, does that mean that Asterisk returned Congestion??? |
| 14:35.36 | Baylink | Very odd, drfreeze |
| 14:36.02 | Baylink | OlafsenM: what kind of channel? |
| 14:36.34 | OlafsenM | SPEECH? |
| 14:36.58 | OlafsenM | Baylink: this is what i'm doing |
| 14:37.15 | Baylink | OlafsenM: PRI, analog, SIP? |
| 14:37.18 | OlafsenM | i'm receiving DAHDI calls and Dialing over IAX |
| 14:37.21 | OlafsenM | PRI |
| 14:37.31 | Baylink | Ok. |
| 14:37.39 | OlafsenM | but, secs after DAHDI accepts call I get "hangup ACK" |
| 14:37.40 | Baylink | And outbound calls are bouncing code 52? |
| 14:37.53 | OlafsenM | and my telco side is getting 52 |
| 14:37.55 | OlafsenM | BUT |
| 14:38.06 | OlafsenM | the call still remains active |
| 14:38.23 | OlafsenM | and it stays active until I manually hang it up |
| 14:38.35 | Baylink | So your problem's not the 52, it's the "doesn't pass supervision to the other side"? |
| 14:39.23 | OlafsenM | my problem is that my ISDN telco somehow gets from my Asterisk box "CONGESTION" signal, but Asterisk still leaves call active |
| 14:39.27 | OlafsenM | and screws my CDR |
| 14:40.37 | Baylink | Yeah; if you place a call from * out a PRI, and get a 52, you should drop all segments of the call. Debug the PRI to a file and see if the 52's coming from the other end. You might want to call the telco and see if they've accidentally reconfigured your switchtype without telling you. |
| 14:41.19 | OlafsenM | well, i'm getting my call FROM PRI to * |
| 14:41.25 | OlafsenM | i'm dialing OUT with IAX |
| 14:41.31 | Baylink | Oh; this is incoming? |
| 14:41.35 | OlafsenM | y |
| 14:41.43 | OlafsenM | that's what makes it even funnier |
| 14:41.47 | Baylink | pb me some logs, willya? |
| 14:41.55 | OlafsenM | well, i can |
| 14:41.59 | Baylink | I don't have enough of my own problems this week. :-) |
| 14:42.36 | [TK]D-Fender | Baylink: Reminds me of an ex who'd say "Got a problem? No? You want one? :D" |
| 14:42.52 | Baylink | Heh. |
| 14:44.32 | *** join/#asterisk decklar (~c41f51be@gateway/web/freenode/x-mbljamrbbkgrlmxl) |
| 14:45.44 | decklar | hey guys - is there an up to date guide on Install Asterisk and Freepbx. (Not asteriskNOW)? |
| 14:46.07 | beek | ~freepbx |
| 14:46.08 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 14:46.47 | decklar | ah ok thanks. Forget about fpbx then. |
| 14:46.56 | decklar | An upto date guide on Asterisk with mISDN? |
| 14:47.03 | decklar | I've found a few |
| 14:47.06 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 14:47.07 | decklar | Yet they are all very different |
| 14:47.25 | decklar | can anyone point me to one they have used and found successful? |
| 14:47.38 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 14:48.51 | [TK]D-Fender | decklar: DAHDI now supports BRI. Forget mISDN |
| 14:49.06 | *** join/#asterisk NiugeS (~NiugeS@mail.cjs-solutions.com) |
| 14:49.30 | OlafsenM | Baylink:http://pastebin.com/T3Zi68pX |
| 14:49.36 | OlafsenM | this is all I have |
| 14:49.39 | decklar | Ok, I will research DAHDI then. Having trouble choosing with tutorial to go with albeit plenty of them! |
| 14:49.53 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 14:50.05 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 14:50.32 | Baylink | OlafsenM: I don't think that's enough log. Is it an active system? |
| 14:52.20 | OlafsenM | yep, sorry |
| 14:52.30 | OlafsenM | some1 screwed up the log levels |
| 14:53.22 | decklar | Btw, can anyone recommend using yum to install asterisk instead of downloading tar's? |
| 14:53.47 | Chainsaw | decklar: You'll be stuck with what Redhat feels are sensible defaults. |
| 14:54.00 | Chainsaw | decklar: This is fine if RedHat and you agree, so you can try it. |
| 14:54.01 | leifmadsen | decklar: I can't recommend that. |
| 14:54.16 | Chainsaw | decklar: But you will likely find that you do not. In which case you'll have to build from source anyway. |
| 14:54.24 | decklar | Ah ok |
| 14:54.46 | Chainsaw | (Distributions like Gentoo offer USE-flags on packages, which fixes most of the problems with this approach. RPMs do not, so they are best avoided) |
| 14:55.03 | *** part/#asterisk OlafsenM (~mark.olaf@193.198.31.85) |
| 14:55.14 | decklar | I best rely on make install from source then. Got it. |
| 14:55.26 | Chainsaw | decklar: *nod* That is leifmadsen-approved method. |
| 14:55.36 | drfreeze | asrt |
| 14:56.17 | decklar | Lesson learnt! |
| 14:56.40 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-74-70.vif.net) |
| 14:57.37 | decklar | May I post a link to a tutorial? Would like to know if it's a good one to follow? |
| 14:58.11 | decklar | Written Jan 20, 2010 |
| 15:00.01 | [TK]D-Fender | decklar: Instructions are in the TARBALL |
| 15:00.32 | decklar | Brilliant, thanks again! |
| 15:00.46 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 15:01.53 | decklar | I'm going to hang around incase I rm *.* -f again and need your help :P |
| 15:01.55 | decklar | Hope you don't mind :P |
| 15:04.27 | *** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.141.34.mtnl.net.in) |
| 15:05.59 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 15:06.03 | vader-- | what are your thoughts on asterisk gui 2.0 versus like freepbx? |
| 15:07.35 | [TK]D-Fender | [11:01]<[TK]D-Fender>[10:55] <vader--> hmm asterisk gui 2.0 looks interesting <- development = dead. Avoid for production |
| 15:07.57 | vader-- | ahh, sorry didn't see that in the other chan |
| 15:08.00 | vader-- | thanks for the heads up |
| 15:09.28 | *** join/#asterisk coppice (~chatzilla@202.64.176.19) |
| 15:10.30 | vader-- | i seen some screen shots of asterisk gui 2.0 and it looked interesting |
| 15:12.06 | vader-- | so it looks like freepbx/tribox/elastix is the main gui competitors |
| 15:13.30 | [TK]D-Fender | vader--: Switchvox, Thirdlane, Scopserv, etc |
| 15:13.40 | vader-- | i meant open source free wise |
| 15:13.45 | [TK]D-Fender | vader--: Because the rest are just FreePBX + varying amounts of crap |
| 15:15.24 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:16.42 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 15:18.44 | *** join/#asterisk hajkym (hajkym@sion.ihrisko.org) |
| 15:19.05 | *** join/#asterisk Lyma (~lyma@unaffiliated/lyma) |
| 15:21.37 | hajkym | hello how can i disable OPTIONS method in asterisk? |
| 15:21.59 | *** join/#asterisk niekvlessert_ (~niekvless@92.70.112.34) |
| 15:22.01 | niekvlessert_ | hello! |
| 15:22.02 | [TK]D-Fender | hajkym: Stop using QUALIFY |
| 15:22.12 | niekvlessert_ | how would i make an agi script not exit non-zero |
| 15:22.46 | hajkym | in sip.conf? |
| 15:22.52 | [TK]D-Fender | hajkym: Clearly |
| 15:22.53 | hajkym | qualify=no? |
| 15:23.05 | hajkym | ok thx |
| 15:24.09 | *** join/#asterisk Z_God (~julius@wlan237073.mobiel.utwente.nl) |
| 15:24.30 | niekvlessert_ | the thing is that my local channel is gone when the agi script exits non-zero |
| 15:25.41 | [TK]D-Fender | niekvlessert_: And you are showing us nothing. |
| 15:25.52 | niekvlessert_ | hold on |
| 15:26.17 | ChannelZ | yeah at least whip off your top. |
| 15:29.25 | niekvlessert_ | - Executing [100@loggedin:1] AGI("Local/op01_200@loggedin-c4cb,1", "test.php") in new stack |
| 15:29.25 | niekvlessert_ | <PROTECTED> |
| 15:29.25 | niekvlessert_ | <PROTECTED> |
| 15:29.25 | niekvlessert_ | <PROTECTED> |
| 15:29.43 | niekvlessert_ | so no is my local gone |
| 15:29.54 | niekvlessert_ | whenever an agi script is used, the local is gone |
| 15:30.13 | [TK]D-Fender | niekvlessert_: PASTEBIN.... and the entire call, and your DIALPLAN |
| 15:30.57 | niekvlessert_ | ok, i will, later |
| 15:31.04 | niekvlessert_ | but i first i will try to explain |
| 15:31.10 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 15:31.23 | niekvlessert_ | when launching a non-existent agi script, it's fine |
| 15:31.33 | niekvlessert_ | the it exits not non-zero |
| 15:31.40 | *** join/#asterisk houms_ (~houms@wsip-70-167-244-115.dc.dc.cox.net) |
| 15:32.07 | niekvlessert_ | but when an empty agi script is there, which does nothing, except return 0 and exit 0 or whatever, the extension is gone |
| 15:32.09 | houms_ | can anyone help me with sound quality issues? it only occurs on incoming calls for the outside |
| 15:32.24 | houms_ | ext to ext and outgoing calls are great |
| 15:32.34 | [TK]D-Fender | niekvlessert_: PASTEBIN <- I dno't trust your code. Do NOT give me a story. |
| 15:32.58 | houms_ | but the sound quality on incoming is staticy and crackly, and there is an echo of your own voice when speaking. though the other person does not hear it at all on their end |
| 15:33.42 | houms_ | we have all aastra phones (2 6753i, 2 6757i, 10 9133i) all have the same issue |
| 15:34.17 | niekvlessert_ | the extensions.conf is 8 kb, the agi file is empty |
| 15:34.33 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 15:34.38 | [TK]D-Fender | niekvlessert_: useless "info". Now show me the call and the code |
| 15:35.06 | carrar | [TK]D-Fender, how about a checksum of that? |
| 15:35.16 | niekvlessert_ | how would you like me to show you the call? like the asterisk log? |
| 15:36.19 | [TK]D-Fender | niekvlessert_: CLI |
| 15:36.44 | *** join/#asterisk Shaaan (Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 15:38.02 | niekvlessert_ | busy, one question, the exit (loggedin, 200, 4), what's 200 and what's 4 |
| 15:39.00 | niekvlessert_ | 200 is extension, 4 is the line of 200 |
| 15:40.36 | houms_ | can anyone help? |
| 15:40.44 | houms_ | or have any ideas where i should be looking |
| 15:40.45 | [TK]D-Fender | niekvlessert_: PASTEBIN <- |
| 15:40.51 | ChannelZ | houms_: what codec are incoming calls using? Is this from an ITSP? |
| 15:41.06 | houms_ | from analog trunk |
| 15:41.19 | [TK]D-Fender | houms_: Maybe you should give a usable description instead of just "outside" |
| 15:41.23 | houms_ | from CLI when i run sip show channels during a call they show ulaw |
| 15:41.36 | houms_ | external phone calls |
| 15:41.51 | houms_ | meaning calls not orignated from internal |
| 15:42.09 | niekvlessert_ | having a hard time getting 8 kb in pastebin... |
| 15:42.20 | ChannelZ | yes but that could mean via SIP/IAX, T1, etc. That's why I asked. |
| 15:42.56 | [TK]D-Fender | ....... |
| 15:43.04 | ChannelZ | You might need to tweak on your hardware, is this a new installation? |
| 15:43.07 | houms_ | analog trunk (ZAP) using openvox a800p with 8fxo ports |
| 15:43.21 | houms_ | yes, we are testing the new system |
| 15:43.23 | [TK]D-Fender | houms_: Where are your phones relative to your server? |
| 15:43.27 | houms_ | <PROTECTED> |
| 15:43.41 | houms_ | they are on same LAN same physical location |
| 15:44.00 | [TK]D-Fender | houms_: What ver of * and what driver? |
| 15:45.03 | houms_ | using trixbox 2.8.0.3 |
| 15:45.20 | houms_ | what is the best way to get you the * ver.? |
| 15:45.41 | houms_ | when you say driver for what ? I apologize for the dumb questions |
| 15:46.37 | houms_ | Asterisk 1.6.0.10-FONCORE-r40 |
| 15:46.45 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-71-190-78-39.nycmny.east.verizon.net) |
| 15:46.56 | [TK]D-Fender | houms_: Zap or Dahdi? "asterisk -rx "show version"" |
| 15:48.18 | niekvlessert_ | http://pastebin.com/XJTgb3QU |
| 15:48.29 | Shaaan | is anyone in here from Orlando, FL? |
| 15:48.31 | *** join/#asterisk houms_ (~houms@wsip-70-167-244-115.dc.dc.cox.net) |
| 15:48.39 | KingDavidNYC | Hello guys, can anybody please suggest me a reason why I would not have audio on meetme if I have ztdummy running on 1.4.30? |
| 15:48.40 | niekvlessert_ | i've pasted extensions.conf, not working cli, working cli |
| 15:49.20 | niekvlessert_ | the working cli is running test2.php, which does not existe |
| 15:49.29 | houms_ | any ideas? |
| 15:49.35 | niekvlessert_ | [TK]D-Fender: here you go :) |
| 15:50.03 | [TK]D-Fender | niekvlessert_: Where does this call come from? |
| 15:50.12 | niekvlessert_ | from a switchboard application |
| 15:52.54 | niekvlessert_ | this should happen: -- Executing [100@loggedin:2] Dial("Local/op01_200@loggedin-d921,1", "SIP/4Z0YfN0X|20") in new stack |
| 15:54.16 | niekvlessert_ | but what happens is this: -- Executing [100@loggedin:2] Dial("SIP/LtjqiXD0-0000010e", "SIP/4Z0YfN0X|20") in new stack |
| 15:54.24 | niekvlessert_ | because of == Spawn extension (loggedin, 200, 4) exited non-zero on 'Local/op01_200@loggedin-2dd3,2' |
| 15:59.13 | [TK]D-Fender | niekvlessert_: I can't tell from all that mes |
| 16:00.29 | niekvlessert_ | you asked for it... |
| 16:00.51 | niekvlessert_ | the thing is, the local channel is gone |
| 16:00.59 | niekvlessert_ | instead of that it starts to use the sip channel |
| 16:02.04 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 16:05.07 | niekvlessert_ | what exactly does == Spawn extension (loggedin, 200, 4) exited non-zero on 'Local/op01_200@loggedin-20c5,2' mean |
| 16:05.39 | niekvlessert_ | exten => _XX.,n,Dial(SIP/${ext},20) |
| 16:05.43 | niekvlessert_ | is the command |
| 16:05.58 | niekvlessert_ | but why does it work when not doing the agi thing!! |
| 16:06.56 | *** join/#asterisk orn (~orn@office.sip.is) |
| 16:08.10 | *** join/#asterisk iscario (~quassel@241.244.71-86.rev.gaoland.net) |
| 16:08.19 | *** join/#asterisk thegoat (~thegoat@c-71-224-155-44.hsd1.pa.comcast.net) |
| 16:08.50 | orn | Whenever I try to use the featurecode attended transfer (as configured in features.conf), I get the playback "Transfer" and a dial tone, but as soon as I enter the first digits, I get "WARNING[11165]: features.c:1436 builtin_atxfer: Did not read data.", which is the same error as occurs in a timeout when no digits are entered. Anyone have any ideas? |
| 16:09.27 | orn | I was thinking maybe dtmfmethod, but I am able to enter the DTMF to activate the feature without any problems... |
| 16:10.26 | p3nguin | kingdavidnyc: Upgrade to dahdi and see if the problem is fixed. |
| 16:11.55 | *** join/#asterisk basty (~basty@2001:4cd8:1:0:21c:b3ff:fec2:ec18) |
| 16:11.57 | basty | Hi |
| 16:13.42 | basty | I have a little problem on my asterisk 1.4.29. I am using dahdi with a digium T1...everything works well beside, that as soon as somebody tries to make a call forward (out of his snom phone) to another internal extension. As soon as you try to call the phone, that uses that call forward..I dont hear any ringing. It seems to be a problem somewhere on the local channel forwarding....is that a known bug in 1.4.29 ? what could I do to solve tha |
| 16:15.21 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 16:15.21 | Naikrovek | the internal extensions are all on the same side of this T1? |
| 16:16.44 | iscario | Hi, |
| 16:16.44 | iscario | I set up an asterisk server that i use with iax accounts. Everything is working fine, but, for personnal reason i need to insert a home made proxy on my server which role is to encrypt data. |
| 16:16.44 | iscario | Here i encountered what i suppose to be a timeout prb : i can register with my account but i can not handle a call... here is the debbug file : http://pastebin.org/301176 |
| 16:24.05 | basty | Naikrovek: well..I have like 120 Snom phones connected to the Asterisk 1.4.29 the outbound is going via T1 (dahdi). Now, if my internal phone "100" has a call forward to another internal phone like "90" and I call for example 069-1234-100 - the call is getting forwarded to SIP/90 - while the call is getting forwarded I cant hear the ringing. |
| 16:24.33 | Naikrovek | basty: does an internal-to-internal call go over the T1 |
| 16:24.41 | basty | Naikrovek: nope |
| 16:25.03 | Naikrovek | okay, so why did you mention the T1 and dahdi - they're not involved in your problem at all |
| 16:25.33 | Naikrovek | nevermind |
| 16:25.37 | Naikrovek | let me clarify |
| 16:25.51 | basty | Naikrovek: Well..because if I try to call internal it works..as soon as sombody calls from external (he comes over the T1 into the asterisk) it doesnt. |
| 16:26.02 | Naikrovek | extension 10 can have forwarding set up to ring extension 20. |
| 16:26.29 | Naikrovek | if someone at extension 50 calls extension 10, it forwards to 20, and they don't hear ringing |
| 16:26.46 | Naikrovek | if someone OUTSIDE calls extension 10, it forwards to 20, but they DO hear the ringing |
| 16:27.45 | basty | Naikrovek: kinda....."if someone at extension 50 calls extension 10, it forwards to 20, and they DO hear ringing" "f someone OUTSIDE calls extension 10, it forwards to 20, but they DONT hear the ringing" |
| 16:27.56 | Naikrovek | okay i got the ringing backwards |
| 16:27.59 | basty | :-) |
| 16:28.24 | Naikrovek | show us the dialplan for the contexts involved and then show us a sip debug of the non-ringing variety |
| 16:28.25 | basty | thats why I talked about the T1 and dahdi...now..I am not sure if it is a problem of the asterisk or dahdi... |
| 16:28.27 | Naikrovek | ~pb |
| 16:28.28 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 16:28.50 | basty | okay sec |
| 16:32.12 | pigpen | Hi all, I am confusing myself, so I figure I'll post here as I have not done "exactly" this before. |
| 16:32.30 | pigpen | Customer called, noted that their old VM system died. (running something on OS2) |
| 16:32.58 | pigpen | So, my suggestion is to use an 8 port digium card in one of their many linux boxes running asterisk. |
| 16:33.01 | iscario | no idea ? |
| 16:33.43 | pigpen | what I am racking my brain on is do I need FXO or FXS ports on the box to receive the VM calls? |
| 16:34.19 | *** join/#asterisk erbse (~erbse@cust.static.109-164-224-215.swisscomdata.ch) |
| 16:34.41 | pigpen | Also, it seems the existing solution can act as an auto-attendant and xfer calls via a "flash" .... does this change the fxo/fxs? |
| 16:37.07 | pigpen | NM....the existing system is producing ring voltage, so I need FXO ports. |
| 16:37.18 | basty | Naikrovek: http://pastebin.com/c9fvT3cy |
| 16:37.20 | pigpen | Don't know about the flash part, but I can google that. |
| 16:38.08 | *** join/#asterisk dms (~dms@173-10-22-217-BusName-utah.ut.hfc.comcastbusiness.net) |
| 16:38.43 | pigpen | cmd Flash, how about that. Google does wonders. |
| 16:38.52 | dms | question for you guys . . |
| 16:38.54 | pigpen | thanks for watching me figure it out myself. |
| 16:38.56 | pigpen | :) |
| 16:39.18 | dms | Been so long since I've been on irc, I can't even remember |
| 16:40.00 | basty | Naikrovek: in this example I dont hear a busy...if I call that external extension...it looks like the call cant get esthablished |
| 16:40.01 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 16:40.11 | DelphiWorld | any asterisk softphone? |
| 16:40.14 | pigpen | dms go ahead. |
| 16:40.20 | DelphiWorld | i mean iax2 phone |
| 16:40.23 | pigpen | DelphiWorld, yeah, zoiper |
| 16:40.24 | DelphiWorld | except of zoiper |
| 16:40.24 | dms | well, this is more of a dev question |
| 16:40.30 | pigpen | too late. |
| 16:40.32 | dms | should I be in asterisk-dev? I suppose |
| 16:40.36 | Naikrovek | basty: alright. something just came up and i can't look at this now, but wait patiently and someone will help |
| 16:40.39 | DelphiWorld | don't need zoiper |
| 16:40.43 | pigpen | dms, depends on the question. |
| 16:40.54 | basty | Naikrovek: okay |
| 16:41.15 | pigpen | DelphiWorld, what are you looking for then? Features wise. |
| 16:41.24 | dms | I've got app_swift just about ready to submit and am wondering about the build process within the asterisk code and where to stick things for the make process to find the swift libs |
| 16:41.50 | DelphiWorld | pigpen: do you know accessibility? |
| 16:41.56 | pigpen | There are some devs in here...but yeah....dev |
| 16:42.05 | dms | thanks :) |
| 16:42.10 | DelphiWorld | pigpen: zoiper is not yet accessible to blind users |
| 16:42.16 | *** part/#asterisk dms (~dms@173-10-22-217-BusName-utah.ut.hfc.comcastbusiness.net) |
| 16:42.30 | pigpen | ah...yeah, I have a friend that is blind. |
| 16:42.44 | *** join/#asterisk mrgabu (~gbdurante@187.38.158.209) |
| 16:43.20 | pigpen | hmm...find a IAXy? |
| 16:43.36 | pigpen | not softphone solution...but use a regular. |
| 16:44.25 | pigpen | does it have to be iax? |
| 16:44.40 | pigpen | I think eyebeam has some options to use a usb handset. |
| 16:44.42 | DelphiWorld | pigpen: yes |
| 16:44.47 | *** join/#asterisk x303 (~X303@20.226.118.70.cfl.res.rr.com) |
| 16:48.41 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 16:48.55 | pigpen | seems there are a few of real iax phones...just google. |
| 16:49.08 | pigpen | otherwise, setup a vpn and get a polycom and run sip. |
| 16:49.14 | mcrownover | afternoon everyone |
| 16:52.28 | [TK]D-Fender | IAXy = suck. All IAX phones on the market... pretty much suck |
| 16:53.19 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:55.59 | pigpen | heh...yeah, I prefer my poly |
| 16:56.49 | pigpen | I have a soekris setup with a nicely packaged gentoo image, doing vpn's for longer term projects so I may have my trusty Polycom 650 with me. |
| 16:57.30 | Corydon76-dig | It's not so much that the IAXy sucks... it's that it's an AT device... |
| 16:57.37 | basty | Mmm..well..anyone else could give me a hand on my problem: "http://pastebin.com/c9fvT3cy" <- Problem is that as soon as somebody is doing a call forward - I cant hear a ringing, busy or playback...(if I call internal - it works - it just happens, when I call the external extension). |
| 16:58.59 | *** join/#asterisk hfb (~hfb@96.247.64.187) |
| 17:02.56 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 17:02.57 | mrgabu | hi all, i am getting some problems with a Digium TE205P , Libpri 1.4.11, DAHDI 2.3.0, Asterisk 1.6.2.8, specially on CALLERID(name) from a CPE link to NETwork side |
| 17:03.13 | *** join/#asterisk aceio (~5d60a88a@gateway/web/freenode/x-ljtdrbhthaufofvm) |
| 17:03.18 | mrgabu | http://pastebin.com/XHJm9X3Z |
| 17:05.20 | [TK]D-Fender | Corydon76-dig: Only provisioned, not even supported any more, chunky... meh |
| 17:06.15 | [TK]D-Fender | - Going to extension s|1 because of immediate=yes <------- BAD |
| 17:06.52 | [TK]D-Fender | mrgabu: also you aren't showing PRI debug |
| 17:06.56 | mrgabu | haha that is not a real scenario, just for testing |
| 17:07.38 | [TK]D-Fender | mrgabu: Well your test is incomplete and is using 1 bad setting right at the start |
| 17:09.37 | *** join/#asterisk reallost1 (~reallost@adsl-065-015-161-123.sip.mem.bellsouth.net) |
| 17:15.02 | *** join/#asterisk astrutt (~astrutt@pinky.ratman.org) |
| 17:24.08 | mrgabu | i've changed some configurations and enabled debug |
| 17:24.24 | mrgabu | if possible, see this link http://pastebin.com/JC9i93t1 thanks in advanced |
| 17:24.43 | *** join/#asterisk MiserySoft (~lnd@83.218.148.46) |
| 17:29.39 | *** join/#asterisk Marquel (~Flinx@p548E7496.dip.t-dialin.net) |
| 17:38.25 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.133.46) |
| 17:45.07 | mrgabu | the variable ${CALLERID(name)} is not delivered from cpe side to network side ? |
| 17:46.48 | Qwell | it's delivered. the provider will just drop it |
| 17:51.58 | *** join/#asterisk diegomad (~mad@190.158.77.2) |
| 17:52.46 | *** part/#asterisk kam187 (~kam187@81-179-8-102.dsl.pipex.com) |
| 17:56.13 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 17:56.59 | guax | i noticed asterisk send to the AMI the event when the channel leaves the queue but i have no event that he entered the queue. does this exist? |
| 17:57.18 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 17:58.29 | mrgabu | Qwell: for example I am making tests with a Digium dual E1 card, 1st as NET and 2nd as CPE... with a RJ48C cable, it is not beeing delivered, just like on my last log |
| 18:01.11 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 18:02.06 | guax | sorry, i found the join |
| 18:05.53 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 18:06.34 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:07.31 | *** part/#asterisk erbse (~erbse@cust.static.109-164-224-215.swisscomdata.ch) |
| 18:09.26 | *** join/#asterisk houms_ (~houms@wsip-70-167-244-115.dc.dc.cox.net) |
| 18:18.21 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 18:18.44 | spiceycurry | I am getting a core dump, and asterisk will not load.... where should I start? |
| 18:18.59 | spiceycurry | ~coredump |
| 18:18.59 | infobot | i guess coredump is the guy who maintains the hentges distribution @ http://www.hentges.net |
| 18:19.14 | leifmadsen | ~backtrace |
| 18:19.14 | infobot | rumour has it, backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt) |
| 18:19.24 | leifmadsen | http://svn.asterisk.org/svn/asterisk/trunk/doc/backtrace.txt |
| 18:19.24 | spiceycurry | ok cool |
| 18:19.25 | spiceycurry | thanks |
| 18:23.01 | *** join/#asterisk reallost1 (~reallost@adsl-065-015-161-123.sip.mem.bellsouth.net) |
| 18:32.35 | spiceycurry | hmm |
| 18:32.36 | spiceycurry | http://www.asterisk.org/downloads/asterisk/1.6.2.8 |
| 18:32.41 | spiceycurry | that link is not workin |
| 18:33.00 | mrgabu | http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 18:33.16 | spiceycurry | ty ty |
| 18:33.59 | spiceycurry | ~love |
| 18:34.00 | infobot | BABY DON'T HURT ME, DON'T HURT ME, NO MORE |
| 18:35.31 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 18:35.43 | [TK]D-Fender | WHAT IS LOVE? |
| 18:36.22 | spiceycurry | you dirty bastards have that god damned song in my head now. |
| 18:37.42 | spiceycurry | where do I put the DONT_OPTIMIZE compileter option when compiling asterisk? |
| 18:37.55 | spiceycurry | found it |
| 18:37.57 | spiceycurry | :O |
| 18:39.23 | spiceycurry | before I blow stuff up, I suppose I really should ask... |
| 18:39.39 | spiceycurry | I compiled asterisk 1.6.2, installed aster fax... |
| 18:39.43 | spiceycurry | Sorry |
| 18:39.49 | spiceycurry | Fax for Asterisk |
| 18:40.12 | spiceycurry | I want to update from 1.6.2.1 to 1.6.2.8 |
| 18:40.17 | spiceycurry | via recompiling.. |
| 18:40.28 | spiceycurry | ./configure ; make ; make install |
| 18:40.30 | spiceycurry | etc |
| 18:40.46 | spiceycurry | should I simply download 1.6.2.8? recompile and install? |
| 18:41.02 | spiceycurry | or will I lose my license info/fax for asterisk |
| 18:41.08 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 18:41.31 | *** join/#asterisk korihor (~korihor@190.205.240.212) |
| 18:47.30 | spiceycurry | I have my backtrace.txt |
| 18:47.49 | spiceycurry | think I can post it at pastebin and get some advice? |
| 18:48.55 | spiceycurry | Can someone give me some advice on this crash http://pastebin.com/G1RqNmh9 |
| 18:49.54 | leifmadsen | spiceycurry: where did you read the link that didn't work before? |
| 18:50.19 | leifmadsen | spiceycurry: license information is stored outside of the source code |
| 18:50.27 | spiceycurry | ok cool, thanks :D |
| 18:50.28 | leifmadsen | spiceycurry: /var/lib/asterisk/licenses or something like that |
| 18:50.47 | leifmadsen | spiceycurry: did you just make up that link before, or did you read it somewhere? Want to make sure we don't have inaccurate documentation. |
| 18:51.10 | *** join/#asterisk dymaxion (~dymaxion@210.177.215.65) |
| 18:51.11 | spiceycurry | no, that is from the latest tarball |
| 18:51.25 | spiceycurry | i.e: go to main asterisk site, and click on start now |
| 18:51.28 | spiceycurry | (for pros's) |
| 18:51.39 | spiceycurry | sorry, go here |
| 18:51.40 | spiceycurry | http://www.asterisk.org/downloads |
| 18:51.52 | spiceycurry | this link is dead: http://www.asterisk.org/downloads/asterisk/1.6.2.8 |
| 18:51.53 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 18:52.11 | leifmadsen | spiceycurry: thanks -- no idea how that got changed like that |
| 18:52.15 | leifmadsen | I will fix it now |
| 18:52.20 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 18:52.27 | spiceycurry | no probs |
| 18:53.06 | leifmadsen | spiceycurry: fixed |
| 18:53.11 | spiceycurry | sweet ty |
| 18:53.42 | spiceycurry | leif, will I need to reinstall fax for asterisk, if I compile the lastest asterisk and install it? |
| 18:53.49 | spiceycurry | or is that config based? |
| 18:53.52 | leifmadsen | spiceycurry: I don't know |
| 18:53.58 | spiceycurry | k, I will give it a shot |
| 18:54.12 | leifmadsen | as long as you don't remove the module in /usr/lib/asterisk/modules/ I don't think so |
| 18:54.19 | spiceycurry | o cool |
| 18:54.22 | spiceycurry | *ok cool |
| 18:54.25 | spiceycurry | thanks agian |
| 18:54.27 | leifmadsen | I would think there should be something about that in the documentation ;) |
| 18:54.57 | spiceycurry | I figured, lol |
| 18:55.29 | spiceycurry | they need a channel called #mom, that just tells you to do what you already know what to do, you just need to be told... again |
| 18:58.04 | *** join/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
| 18:58.12 | niekvlessert | is there any way to slow down ami? |
| 18:58.52 | leifmadsen | slow down? |
| 18:59.27 | niekvlessert | yeah |
| 18:59.33 | leifmadsen | don't think so |
| 18:59.47 | niekvlessert | because my ami command seems to finish before the dialplan is done |
| 18:59.56 | niekvlessert | so it becomes a mess |
| 19:00.09 | niekvlessert | i'm using originate |
| 19:00.14 | niekvlessert | this uses locals |
| 19:00.28 | niekvlessert | but the local is gone before the second dial command has been made |
| 19:04.16 | *** join/#asterisk houms_ (~houms@wsip-70-167-244-115.dc.dc.cox.net) |
| 19:04.42 | houms_ | does anyone know if you can change the ringtone for a transferred call so it is different from the incoming ringtone? |
| 19:04.48 | houms_ | for aastra phones? |
| 19:06.26 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 19:07.24 | [TK]D-Fender | houms_: You can only detect a blind transfer with a special channel var that egts set. From there do whatever you want. |
| 19:09.02 | houms_ | can you please clarify what you mean by egts? |
| 19:09.34 | [TK]D-Fender | gets |
| 19:09.36 | leifmadsen | houms_: it was a typo, you meant "gets" |
| 19:10.03 | leifmadsen | s/you/he |
| 19:10.24 | houms_ | currently incoming calls come in with a ring and when I want to tranfser to another extension, it rings the same way as a incoming call. |
| 19:10.47 | [TK]D-Fender | houms_: Already answered. |
| 19:10.52 | houms_ | if i read your response correctly, your saying i need to first set a special variable to detect the transfer first before trying to change the tone? |
| 19:11.12 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-74-70.vif.net) |
| 19:11.23 | houms_ | is there any documentation that can explain it a little more in depth? |
| 19:11.33 | carrar | hahah http://www.deansproperty.com.au/Home/Profiles |
| 19:11.35 | leifmadsen | houms_: http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
| 19:11.59 | houms_ | thanks leifmadsen, i assume it applies to aastra phones as well? |
| 19:12.09 | leifmadsen | it may or it may not |
| 19:12.25 | [TK]D-Fender | houms_: There is a page on the wiki describing Aastra distinctive ringing as well |
| 19:12.30 | [TK]D-Fender | houms_: Go look |
| 19:13.47 | niekvlessert | houms_: add dr4 belcore or something sipheader whenever you like |
| 19:14.13 | niekvlessert | works fine for me |
| 19:14.20 | houms_ | niekvlessert what do you mean? in the webui or aastra/mac.cfg? |
| 19:14.32 | niekvlessert | in extensions.conf |
| 19:15.24 | niekvlessert | [TK]D-Fender |
| 19:15.26 | niekvlessert | hello! |
| 19:15.33 | houms_ | do you mind posting an example of extensions.conf |
| 19:15.34 | niekvlessert | i'm still looking into my problem |
| 19:15.34 | houms_ | ? |
| 19:15.43 | *** part/#asterisk mrgabu (~gbdurante@187.38.158.209) |
| 19:16.31 | [TK]D-Fender | niekvlessert: I can't concentrate on yours right now, sorry |
| 19:18.35 | niekvlessert | 1234,n,SIPAddHeader(Alert-Info: info=<Bellcore-dr4>) |
| 19:18.37 | niekvlessert | ok |
| 19:18.40 | niekvlessert | no problem |
| 19:19.07 | niekvlessert | who's the ami guru here :) |
| 19:19.10 | *** join/#asterisk x303 (~X303@20.226.118.70.cfl.res.rr.com) |
| 19:19.29 | p3nguin | Is there any way to output the values of every variable that has a value set on it? Similar to the system shell's "env" command. |
| 19:19.58 | houms_ | niekvlessert , what does that do exactly? it seems like it rings incoming calls from the outside differently? is that correct? |
| 19:20.30 | niekvlessert | well depends on the dialplan |
| 19:20.41 | niekvlessert | you can add this any place in the dialplan |
| 19:20.44 | p3nguin | If the a phone supports distinctive ringing, you could certainly use it for that. |
| 19:20.46 | niekvlessert | so also when doing a transfer i guess |
| 19:21.09 | leifmadsen | p3nguin: I think you want DumpChan() |
| 19:21.41 | p3nguin | leifmadsen: I think so, too. Thanks. |
| 19:22.07 | houms_ | well all of our calls go to ring group. |
| 19:22.54 | houms_ | depending on who answers the call needs to be transferred to another party inhouse, so in transferring we would like to have the ring of the transfer be different then the original call ring was |
| 19:23.26 | houms_ | as the person we are transferring does not know if the call is a transfer or not |
| 19:24.21 | [TK]D-Fender | houms_: As I said there is a special VARIABLE set when you do a BLIND TRANSFER. Go check for it before you dial your phones and set the ringer accordingly <- |
| 19:26.56 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 19:27.34 | exothermc | When doing PRI termination in north america do you typically need echo cancelation on the T1 cards? |
| 19:27.52 | [TK]D-Fender | exothermc: yes |
| 19:27.56 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 19:28.32 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.137.237.dsl.dyn.forthnet.gr) |
| 19:28.36 | exothermc | What scenario do you not? Local market etc? |
| 19:29.15 | [TK]D-Fender | exothermc: In just about all |
| 19:29.20 | [TK]D-Fender | (needed) |
| 19:29.40 | exothermc | damn |
| 19:29.46 | exothermc | They don't come cheap |
| 19:30.09 | coppice | exothermc: if you only call very long distances from a digital trunk, someone else will deal with the far end echo for you :-) |
| 19:30.24 | exothermc | ahh yes |
| 19:32.04 | leifmadsen | p3nguin: ya, it's super useful for debugging |
| 19:33.05 | leifmadsen | houms_: if you're doing attended transfers from the SIP phone, Asterisk won't know it is a transfer while executing dialplan, so you won't be able to do that from the dialplan in that scenario |
| 19:35.29 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 19:37.09 | houms_ | leifmadsen do you have any suggestions how to achieve it? i am confused as you referred to our setup as using attended transfers but earlier someone referred to them as blind transfers |
| 19:37.30 | leifmadsen | houms_: blind and attended transfers are different things |
| 19:37.52 | leifmadsen | houms_: you just say, "transferring" so it depends what kind of transfer you're doing and how they are being achieved. |
| 19:37.59 | houms_ | right and in trying to understand the difference and what we are using I am a little confused |
| 19:38.03 | houms_ | fair enough |
| 19:39.05 | leifmadsen | ~blind-transfer |
| 19:39.09 | houms_ | if i get a call, and they want to speak to someone else, i will place them on hold, and either intercom or call before transferring, |
| 19:39.13 | [TK]D-Fender | houms_: When you do an attended transfer it is a completely new and separate call from that phone. Its only when you confirm the call to be handed off does the phone tell * to connect A to B |
| 19:39.16 | houms_ | what is this considered technically? |
| 19:39.28 | [TK]D-Fender | houms_: Therefor * has no idea of your intentions as it starts processing the call |
| 19:39.33 | leifmadsen | right |
| 19:39.38 | leifmadsen | it just looks like a call |
| 19:39.45 | [TK]D-Fender | [15:39]<houms_>if i get a call, and they want to speak to someone else, i will place them on hold, and either intercom or call before transferring, <- VAGUE. |
| 19:40.03 | leifmadsen | sounds like an attended transfer though |
| 19:40.13 | [TK]D-Fender | houms_: Actions taken aside from the transfer have nothing to do with the transfer itself. |
| 19:40.19 | leifmadsen | at which point it is not really a transfer until the transfer actually happens |
| 19:40.35 | houms_ | if they want the call i pick up line 1 (original call) and hit xfer +ext then xfer again |
| 19:40.39 | [TK]D-Fender | If I make a sandwich before I transfer... is it "blind" then? |
| 19:41.00 | vader-- | what is the command to reload all the conf files? it's nolonger reload like 1.2 |
| 19:41.00 | [TK]D-Fender | houms_: Depends what kind of transfer that is on your phone. Go read its manual |
| 19:42.15 | houms_ | aastra saysit does blind or consultative |
| 19:42.45 | houms_ | Transfer key - Allows you to perform blind or consultative transfer of an |
| 19:42.46 | houms_ | active call to another number. In blind transfer, you transfer the call without |
| 19:42.47 | houms_ | waiting for the far end to answer. In consultative transfer, you wait for the far |
| 19:42.49 | houms_ | end to answer before transferring the call. |
| 19:42.53 | [TK]D-Fender | houms_: And what does it says your actions represent? |
| 19:42.56 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 19:43.18 | houms_ | blind transfer since i am not waiting til the call is answered correct? |
| 19:43.29 | [TK]D-Fender | houms_: Go try |
| 19:43.53 | p3nguin | vader--: <something> reload |
| 19:44.04 | p3nguin | vader--: dialplan reload, sip reload, et cetera |
| 19:45.03 | *** join/#asterisk neurosys (~neurosys@67.32.132.60) |
| 19:46.38 | *** join/#asterisk houms_ (~houms@wsip-70-167-244-115.dc.dc.cox.net) |
| 19:46.54 | houms_ | okay, based on that it is not a blind trasnfer its consultative |
| 19:48.16 | houms_ | again confusing since the aastra doc says the default value is blind transfer is disabled |
| 19:48.32 | houms_ | and since i do not see the option set in either mac.cfg or aastra.cfg |
| 19:50.02 | *** join/#asterisk MhaddogM1 (~MhaddogM1@173-109-47-24.pools.spcsdns.net) |
| 19:52.06 | spiceycurry | I am getting "chan_local.c:534 local_call: No such extension/context 9052983702@outboundialcontext while calling Local channel" --- what am I missing? |
| 19:52.43 | [TK]D-Fender | spiceycurry: And extension to match 9052983702 in [outboundialcontext] |
| 19:52.54 | spiceycurry | ok, thanks |
| 19:54.29 | *** join/#asterisk rjk` (rjk@home.melb.org) |
| 19:55.07 | rjk` | I have done alot of googling , can anyone suggest a nice php based interface for asterisk as I am running it on a mac |
| 19:55.38 | p3nguin | Ah, the mysterious wonders of PLAIN-FUCKING-TEXT error messages. Gotta love them. |
| 19:57.00 | [TK]D-Fender | p3nguin: Doctor, Doctor... it hurts when I raise my arm like... awwwww FUKKIT |
| 19:57.41 | [TK]D-Fender | rjk`: You were looking to do the automated call-outs, right? |
| 19:58.37 | carrar | CLI is where it's at! |
| 19:58.42 | niekvlessert | guys, can you please look here? http://pastebin.com/zW0L5sjY |
| 19:58.49 | niekvlessert | it's about ami |
| 19:59.46 | Trixboxer | Hi, How can I see current call time in CLI ? |
| 19:59.59 | carrar | 11-1 is ringingerbose |
| 20:00.01 | carrar | err |
| 20:00.09 | carrar | show channels verbose |
| 20:00.15 | p3nguin | core show channels concise |
| 20:00.35 | p3nguin | or core show channels verbose |
| 20:01.31 | spiceycurry | Does anyone know a good tutorial or just a plain "how-to" for "Fax for Asterisk" ? The docs aren't too great |
| 20:01.31 | Trixboxer | p3nguin, Thnx ;) |
| 20:03.44 | niekvlessert | http://www.digium.com/en/products/software/faxforasterisk.php |
| 20:04.06 | spiceycurry | I am referring to those |
| 20:04.27 | spiceycurry | is there a better resource? with examples |
| 20:04.32 | niekvlessert | what's wrong with the docs |
| 20:04.35 | niekvlessert | they're good |
| 20:05.41 | spiceycurry | I saw the example extensions.conf |
| 20:05.59 | spiceycurry | is that all that needs to be set, exactly as it is to receive a fax? |
| 20:06.08 | spiceycurry | or is there something I missed |
| 20:06.30 | niekvlessert | do you want t-38 licensed? |
| 20:06.40 | spiceycurry | yes |
| 20:07.08 | *** join/#asterisk CraigW76_ (~techcaw@addr33.mimc.com) |
| 20:07.15 | spiceycurry | I have 10 channels that I purchased, as well as 1 free one.... I am using T.38 |
| 20:07.47 | spiceycurry | (my fax show stats is perfect) |
| 20:08.56 | coppice | why didn't you get the free one working first? |
| 20:08.56 | spiceycurry | doesn't matter, we are using it regardless, lol |
| 20:08.57 | spiceycurry | I only do what I am told lol |
| 20:09.13 | [TK]D-Fender | spiceycurry: Jump off the bridge! |
| 20:09.43 | spiceycurry | believe me, I want to lol |
| 20:10.04 | spiceycurry | we're using the 10 to test, than buying 200 more likely |
| 20:10.55 | *** join/#asterisk {Repelex} (~{Repelex}@201-71-129-252-arpa.vsp.com.br) |
| 20:10.57 | spiceycurry | All I want to get working is a simple line to pick up, and save the file to a directory. |
| 20:11.07 | spiceycurry | I'd even be happy if I could get a fax sent out |
| 20:11.18 | coppice | so, someone does actually buy it :-) |
| 20:11.21 | spiceycurry | lol |
| 20:11.42 | vstemen | Is part of asterisk written in fortran? |
| 20:12.48 | spiceycurry | Can someone paste a link to an extensions.conf file that will work to receive a fax? Or post a good link in order to set it up for a simpel test? |
| 20:12.53 | coppice | quite a few people seem to run the one free channel of FFA, but big users seem to go from a free option |
| 20:13.50 | spiceycurry | I'd be happy to get a tutorial on setting the free one up.. |
| 20:14.47 | ChannelZ | the docs are not that complicated |
| 20:15.21 | spiceycurry | well, I have the extensions config setup exactly as their docs do... |
| 20:15.34 | spiceycurry | <PROTECTED> |
| 20:15.36 | vstemen | I was just wondering why the BSD port requires f2c, the Fortran to C compiler |
| 20:15.58 | [TK]D-Fender | spiceycurry: It s **1** line of dialplan. Holy f-ing crap. |
| 20:16.05 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 20:16.31 | spiceycurry | I added it, reloaded everything |
| 20:16.34 | ChannelZ | exten => 1,1,ReceiveFax(/tmp/barf.tif) |
| 20:16.41 | ChannelZ | Dial 1.. |
| 20:16.46 | ChannelZ | shrugs |
| 20:16.49 | rjk` | [TK]D-Fender: succeeded in doing the project, now looking for some PHP interfaces |
| 20:16.51 | spiceycurry | is there perhaps another file I need to say? |
| 20:16.58 | [TK]D-Fender | spiceycurry: And are showing us precisely NOTHING |
| 20:16.58 | spiceycurry | I mean modify? |
| 20:17.04 | grungies1138 | ok, anyone wanna help me tackle a MOH issue? |
| 20:17.05 | [TK]D-Fender | spiceycurry: No. |
| 20:17.17 | [TK]D-Fender | rjk`: What are you expecting this to do? |
| 20:17.33 | spiceycurry | so I dont need to modify or configure my sips.comf? |
| 20:17.35 | ChannelZ | spiceycurry: you've already installed the fax modules and license? |
| 20:17.36 | spiceycurry | *conf? |
| 20:17.41 | spiceycurry | Everything is installed |
| 20:17.47 | rjk` | [TK]D-Fender: help me train someone in using Asterisk without CLI for when I am not available |
| 20:18.17 | spiceycurry | http://pastebin.com/WDysCrX8 |
| 20:18.34 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 20:18.34 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 20:18.37 | cusco | hi |
| 20:18.47 | pabelanger | grungies1138: Don't ask for help, just ask your question |
| 20:19.03 | [TK]D-Fender | rjk`: Too vague. And all of them completely take over the configuration of your system. Which means your previous work is trashed |
| 20:19.07 | cusco | setting canreinvite=yes (or directmedia=yes) seems to make one asterisk dialing to the next one trough sip fail... |
| 20:19.16 | cusco | handle_response_invite: Failed to authenticate on INVITE to '"932485457" <sip:932485457@192.168.2.7>;tag=as6fe089b2' |
| 20:19.22 | lordvadr | Does the subscribecontext in sip.conf tell voicemail subscriptions where to look, or should the mailbox line have an @context after the mailbox? |
| 20:19.39 | [TK]D-Fender | cusco: medai has nothing to do with auth failures |
| 20:19.54 | [TK]D-Fender | lordvadr: No, that is for hints |
| 20:20.01 | cusco | [TK]D-Fender: I also thought so, but on google I found out somebody posting otherwise... |
| 20:20.06 | grungies1138 | Normally I would Pabelanger, but I did that three times yesterday with no responses. |
| 20:20.07 | cusco | http://lists.digium.com/pipermail/asterisk-users/2007-March/183525.html |
| 20:20.12 | lordvadr | That's what I thought. So box@context in mailbox=? |
| 20:20.25 | [TK]D-Fender | spiceycurry: I see no attempt, no failure, and no code |
| 20:20.30 | ChannelZ | spiceycurry: so what is your actual problem? failures? You can't figure out how to send a fax call to the right extension in your dialplan, or what? |
| 20:20.34 | [TK]D-Fender | lordvadr: Correct |
| 20:20.39 | lordvadr | thank you much! |
| 20:20.42 | grungies1138 | Do you know what format of WAV asterisk will read natively? |
| 20:20.56 | grungies1138 | I tried to convert some mp3s and they are not being read |
| 20:21.06 | cusco | basically Dial(SIP/150@192.168.2.5) should work |
| 20:21.12 | ChannelZ | 16bit or 8bit at 8kHz |
| 20:21.21 | cusco | there is a registration with that ip |
| 20:21.27 | ChannelZ | Mono |
| 20:21.31 | grungies1138 | I also tried just doing the mp3 read with mpg123. No errors but no audio either |
| 20:21.32 | ChannelZ | I always forget to say mono |
| 20:21.45 | spiceycurry | Well, ChannelZ: http://pastebin.com/padGYVew |
| 20:21.50 | spiceycurry | Thats my conf |
| 20:22.16 | bmoraca_work | cusco: is the registration for that callerid? can you tell the other box to use the provided auth credentials, rather than the From: header? |
| 20:22.24 | spiceycurry | I think I should start over... |
| 20:22.52 | rjk` | [TK]D-Fender: humour me, which is the better of the various PHP interfaces in your opinion ? |
| 20:22.53 | pabelanger | grungies1138: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 20:23.11 | [TK]D-Fender | rjk`: Scopserv |
| 20:23.25 | rjk` | cheers |
| 20:23.33 | spiceycurry | If my context is set to default... how would FFA know about the [fax-tx] and [fax-rx] ? |
| 20:23.45 | ChannelZ | spiceycurry: your incoming calls are strictly voip? (sip?) |
| 20:23.45 | cusco | bmoraca_work: you lost me, hold let me paste some info |
| 20:24.00 | bmoraca_work | cusco: is 192.168.2.5 an asterisk box? |
| 20:24.07 | spiceycurry | yes |
| 20:24.12 | spiceycurry | ChannelZ: yes |
| 20:24.14 | *** join/#asterisk bent_screwdriver (~socain00@173.7.82.240) |
| 20:24.15 | ChannelZ | spiceycurry: the way you are setup the 'fax' extension does a Goto into the fax-rx context. But you have no way to even get to that extension |
| 20:25.02 | bmoraca_work | cusco: set match_auth_username=yes in global of sip.conf of the second asterisk box. |
| 20:25.03 | ChannelZ | spiceycurry: unless you have 'faxdetect=yes' set in sip.conf |
| 20:25.10 | spiceycurry | I will check |
| 20:25.15 | ChannelZ | and your incoming calls are winding up in the default context |
| 20:26.18 | spiceycurry | faxdetect = yes <- in my sip |
| 20:26.34 | grungies1138 | So I can just do file convert Blah.mp3 Blah.wav? |
| 20:27.42 | cusco | bmoraca_work: yes it is an asterisk box |
| 20:27.47 | cusco | ok I will try that |
| 20:27.56 | ChannelZ | spiceycurry: ok so does your incoming calls go into the 'default' context, and do you have some sort of dialplan that actually answers the call? |
| 20:28.28 | spiceycurry | I dont have a dial plan to answer the call |
| 20:28.38 | spiceycurry | unless its in that extensions.conf |
| 20:28.44 | ChannelZ | Ok, so you expect to be able to detect an incoming fax call how? |
| 20:29.03 | spiceycurry | Would the dial plan entry go into extensions.comf? |
| 20:29.08 | spiceycurry | *conf |
| 20:29.11 | ChannelZ | yes |
| 20:29.36 | [TK]D-Fender | OMG |
| 20:29.58 | cusco | bmoraca_work: no difference: [Jun 2 21:29:41] NOTICE[6202]: chan_sip.c:17767 handle_response_invite: Failed to authenticate on INVITE to '"932485457" <sip:932485457@192.168.2.7>;tag=as57046b84' |
| 20:30.04 | ChannelZ | Is this a totally new system? You bought 11 channels of fax and have ITSP service but aren't even setup to answer basic calls? I'm confused |
| 20:30.19 | spiceycurry | Yea its new |
| 20:30.20 | bmoraca_work | cusco: you're going to need to give more detail then, including sip debugs |
| 20:30.36 | cusco | bmoraca_work: wortking on that, here is box1: http://paste.debian.net/75888/ |
| 20:30.41 | cusco | hold on for box2 |
| 20:30.41 | rjk` | [TK]D-Fender: the config i managed to put together was asterisk from macports, teleyapper 3.0 and the digium skype module |
| 20:30.54 | spiceycurry | it probably should have said this in the fax docs... |
| 20:31.11 | spiceycurry | so I need to setup asterisk to answer calls, then, configure fax? |
| 20:31.19 | [TK]D-Fender | rjk`: Oh... IIRC none of the GUI's support SkyForAsterisk normally... you MIGHT get lucky and be able to hack that in though |
| 20:31.26 | [TK]D-Fender | rjk`: Expect an uphill battle |
| 20:31.41 | [TK]D-Fender | [16:30]<spiceycurry>it probably should have said this in the fax docs... <- No, it shouldn't. |
| 20:31.42 | spiceycurry | a full example setup would have been nice I guess |
| 20:31.51 | *** join/#asterisk Sheeplet (~Fallen@dsl-246-127-24.telkomadsl.co.za) |
| 20:31.53 | grungies1138 | ok file convert can't read mp3 |
| 20:32.05 | Sheeplet | hi all |
| 20:32.07 | rjk` | [TK]D-Fender: i enjoy challenges :) |
| 20:32.14 | [TK]D-Fender | spiceycurry: thats why a car stereo you buy at a store doesn't come with a "How to driev a car" handbook |
| 20:32.15 | ChannelZ | ~book |
| 20:32.15 | 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 |
| 20:32.20 | ChannelZ | spiceycurry: ^^^^ READ |
| 20:32.26 | spiceycurry | Ok, will do :( |
| 20:32.50 | [TK]D-Fender | spiIf you can't understand how to process a call... well then just like that stereo buyer... don't buy the stereo if you can't even drive the damn car |
| 20:33.07 | spiceycurry | so before I read... once I get asterisk answering a phone call... it shouldn't be to hard than, and things will start making sense, right? |
| 20:33.13 | ChannelZ | Asterisk is not just a fax machine.. they don't tell you how to configure an entire PBX in the docs for a single fax module |
| 20:33.22 | [TK]D-Fender | spiceycurry: For most users, yes |
| 20:33.24 | rjk` | [TK]D-Fender: someone answered me on an email list about a thing called Incredible PBX, do you know anything about that ? |
| 20:33.25 | ChannelZ | One would hope |
| 20:33.26 | spiceycurry | ok cool |
| 20:33.38 | spiceycurry | thanks |
| 20:33.44 | [TK]D-Fender | rjk`: Saw the name. Sounded gay. |
| 20:33.56 | rjk` | thats what i thought lol |
| 20:34.11 | ChannelZ | It's SUPER! |
| 20:34.20 | ChannelZ | (i mean SssssUPER!) |
| 20:34.26 | [TK]D-Fender | They would have called it MagicalPBX or WonderfulPBX, but Steve Jobs would haev sued them. |
| 20:34.59 | ChannelZ | hehe |
| 20:35.07 | ChannelZ | I hate that new iPad commercial. It's so fucking pretentious |
| 20:35.21 | [TK]D-Fender | ChannelZ: RDF <- |
| 20:35.24 | ChannelZ | "iPad is thin. iPad is beatiful. iPad is magical." BARF |
| 20:36.01 | cusco | bmoraca_work: box2 http://paste.debian.net/75889/ |
| 20:37.12 | coppice | you missed "iPad is too heavy to hold, but has no stand" |
| 20:38.56 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 20:39.50 | bmoraca_work | cusco: i'm not seeing it attempt to invite after the inital failure, which is what should happen. also, i'm not seeing the peer information that should be in sip.conf. |
| 20:39.53 | grungies1138 | Did you guys see on Tosh.0 where they sent him a free iPad. He used it as a golfball with his 9 iron. |
| 20:40.10 | niekvlessert | hen doing originate in AMI, is there a way to influence how long the local channel will ring? |
| 20:40.13 | spiceycurry | Ok, I am going through the book. My question is, I have 3 ethernet cables going into my digium card- does this mean I will need to configure 3 spans? |
| 20:40.15 | niekvlessert | when doing originate in AMI, is there a way to influence how long the local channel will |
| 20:40.30 | spiceycurry | I have 1 span configured, and it is showing OK in the ALARMS of Dahdi_tool |
| 20:41.43 | bmoraca_work | Nerd Vittles are a bunch of idiots. their writeups are all worthless. |
| 20:42.31 | [TK]D-Fender | Vittles & bits & bits & bits! |
| 20:42.32 | cusco | bmoraca_work: http://paste.debian.net/75892/ that has peer information |
| 20:42.42 | cusco | bmoraca_work: invite after initial failure on box2 you mean? |
| 20:42.42 | bmoraca_work | spiceycurry: you create one span for each port and then use a dahdi group to group them together |
| 20:42.52 | cusco | is it missing some sip debugging info? |
| 20:42.58 | bmoraca_work | cusco: yes. |
| 20:42.59 | spiceycurry | ok cool |
| 20:43.02 | spiceycurry | thanks |
| 20:43.03 | bmoraca_work | cusco: no |
| 20:43.10 | ChannelZ | I'm not trying to be rude.. but I'm curious as to your situation, you now have 3 PRI lines and just setting up Asterisk for the first time ever? |
| 20:43.13 | cusco | maybe I cut it short, because there is always too much info on box2 |
| 20:43.15 | bmoraca_work | cusco: it's not missing debug info...it's missing a step |
| 20:43.38 | cusco | thats what it printed out... |
| 20:43.44 | spiceycurry | ChannelZ: I am a C programmer here... the guy who setup all this stuff left over night |
| 20:43.45 | bmoraca_work | cusco: any particular reason why you're using "host=dynamic" on these? |
| 20:43.53 | spiceycurry | I am stuck with it |
| 20:43.55 | spiceycurry | :( |
| 20:44.00 | ChannelZ | spiceycurry: or you mean the guy who DIDN'T set all this up |
| 20:44.08 | spiceycurry | well, yes that is correct, lol |
| 20:44.24 | spiceycurry | I took a quick look at this a month ago... |
| 20:44.36 | spiceycurry | said fuck it, and took another project... |
| 20:44.55 | cusco | bmoraca_work: I had in the past, its not needed now. Does it make a difference? |
| 20:44.59 | ChannelZ | Maybe your company should just hire a consultant for a day to get the thing off the ground |
| 20:45.05 | spiceycurry | the only thing I know about phones, is I am getting cancer from my iphone |
| 20:45.07 | cusco | bmoraca_work: 2nd attempt at sip debug on box2: |
| 20:45.09 | cusco | http://paste.debian.net/75893/ |
| 20:45.23 | spiceycurry | I am sure I will figure this out with some reading and pissing around |
| 20:45.31 | [TK]D-Fender | spiceycurry: That comes bundled with the "lifetime" warranty ;) |
| 20:45.41 | spiceycurry | haha that is true |
| 20:46.09 | rrb3942 | cusco: after the 401 you should a second invite should be sent that contains the auth headers, which is not happening (asterisk is just giving up for some reason) |
| 20:46.13 | ChannelZ | Yes but you're starting a little backwards, having tried to get fax to work first. It'd be more helpful to have someone who knows what they are doing even come in for a couple of hours and give you a big crash-course |
| 20:46.30 | rrb3942 | cusco: have you tried setting 'alwaysauthreject = no' in sip.conf on the remote system? |
| 20:46.46 | [TK]D-Fender | spiceycurry: there is no "getting fax to work first" |
| 20:46.51 | cusco | rrb3942: I will try... |
| 20:47.01 | spiceycurry | I was hoping lol |
| 20:47.07 | cusco | so this is not somehow related to canreinvite=yes |
| 20:47.09 | cusco | ? |
| 20:47.19 | ChannelZ | It isn't that hard once you have a bigger picture of the whole thing, but you're going to burn out a lot of help you might get here by asking questions every single step of the way |
| 20:47.21 | [TK]D-Fender | spiceycurry: You need to get your devices capable of sending calls to *. You then need a dialplan to process the calls. In that dialplan you will choose when to receive a fax |
| 20:47.32 | [TK]D-Fender | BBIAB |
| 20:48.13 | rrb3942 | cusco: dunno, just saying what you should be seeing, not sure why your not |
| 20:48.21 | rrb3942 | you're* |
| 20:48.33 | cusco | makes no change rrb3942 |
| 20:49.27 | rrb3942 | did the trace change at all? |
| 20:49.39 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 20:50.46 | cusco | rrb3942: http://paste.debian.net/75894/ |
| 20:50.47 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 20:51.50 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 20:53.03 | bmoraca_work | cusco: try specifying the host directly and referencing the peer in your dial by peer name. |
| 20:53.11 | cusco | ok |
| 20:54.20 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 20:54.40 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:55.16 | cusco | weird now I have: chan_sip.c:17779 handle_response_invite: Received response: "Forbidden" from '"932485457" <sip:932485457@192.168.2.7>;tag=as3eae7255 |
| 20:56.12 | lordvadr | If I send a call to a real-time switch from a dialplan context, and nothing is found in that switch (table) that matches, what happens? |
| 20:56.50 | *** join/#asterisk miamiseb (~deigo@208.76.35.132) |
| 20:56.59 | cusco | bmoraca_work: rrb3942 box2 sip debug seems the same: http://paste.debian.net/75895/ |
| 20:58.49 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 21:00.23 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:01.06 | spiceycurry | Is there a dchan on each span? |
| 21:02.35 | *** join/#asterisk sahafeez_ (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 21:02.50 | ChannelZ | yes |
| 21:05.19 | rrb3942 | cusco: does it work if you use hard coded IP address in 'hosts=' and don't use any registrations or authentication? |
| 21:06.32 | rrb3942 | cusco: and for the dial string something like SIP/primario/numbertodial |
| 21:06.33 | cusco | rrb3942: host=blah is that in the peer ? that last paste included it |
| 21:06.41 | cusco | let me try |
| 21:06.59 | rrb3942 | cusco: but you have host=dynamic right now don't you? |
| 21:07.32 | cusco | I changed it |
| 21:07.34 | cusco | still chan_sip.c:17779 handle_response_invite: Received response: "Forbidden" from '"932485457" <sip:932485457@192.168.2.7>;tag=as4d4ad584' |
| 21:07.50 | cusco | with SIP/primario/150 |
| 21:09.03 | cusco | thing is between box2 and box3 it works just with Dial(SIP/${EXTEN:1:3}@10.100.100.3,30,rg); |
| 21:09.31 | rrb3942 | yeah you have something strange going on |
| 21:09.35 | spiceycurry | does this system.conf look valid for 4 port, 3 cables (t1) http://pastebin.com/Ax2H3Gsq |
| 21:09.47 | spiceycurry | I knio wyou won't all know forsure about the fine details... |
| 21:09.54 | spiceycurry | but does that look normal? |
| 21:10.01 | rrb3942 | when you get the Forbidden line, is box2 sending back a 403 forbidden in the trace? |
| 21:10.15 | cusco | let me check |
| 21:10.17 | spiceycurry | *I know you wont all know about the fine details |
| 21:10.31 | rrb3942 | cusco: yeah, it is, I see it in the last trace you posted |
| 21:10.34 | cusco | rrb3942: SIP/2.0 403 Forbidden |
| 21:10.40 | cusco | yes line 123 from paste |
| 21:11.44 | rrb3942 | can you set your logging level to verbose and post the tail of the log while trying a call? |
| 21:11.56 | cusco | oh I just noticed sip show registry on box2: 192.168.2.7:5060 N primario 105 No Authentication Wed, 02 Jun 2010 21:53:59 |
| 21:12.01 | cusco | no authentication ? |
| 21:12.36 | cusco | rrb3942: verbose is set to 15 on both machines |
| 21:12.53 | cusco | Jun 2 22:12:36] WARNING[26041]: chan_sip.c:18080 handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'primario' to '192.168.2.7 |
| 21:12.57 | cusco | hold on, something is wrong |
| 21:13.08 | *** join/#asterisk Benwa (~Benwa@213.219.188.210.adsl.dyn.edpnet.net) |
| 21:13.18 | spiceycurry | this look ok? http://pastebin.com/Ax2H3Gsq |
| 21:14.08 | ChannelZ | probably.. dahdi_genconf does a pretty good job guessing about things most the time |
| 21:14.15 | cusco | err, since I changed host=192.168.2.5 |
| 21:14.18 | cusco | I can't register? |
| 21:14.20 | spiceycurry | shit. lol- I will try that |
| 21:14.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:14.46 | cusco | why? |
| 21:15.12 | ChannelZ | spiceycurry: dahdi_genconf didn't generate what you pasted? (did you just get that someplace else?) |
| 21:15.25 | spiceycurry | I made it from the documentation |
| 21:15.30 | ChannelZ | The only thing I did just notice is the echo canceller is only turned on the first span of channels |
| 21:15.32 | cusco | secret is the same |
| 21:15.37 | cusco | I am not understanding |
| 21:15.52 | spiceycurry | I just ran that tool, it made a nice one... however- there is an extra span, I think I should remark out as I only have 3 cables |
| 21:16.07 | ChannelZ | it's a 4-port card though yes? |
| 21:16.10 | spiceycurry | yes |
| 21:16.20 | spiceycurry | should I leave all 4 generated ones uncommented? |
| 21:16.28 | cusco | it should show as UNCONFIGURED then, no problem spiceycurry |
| 21:16.28 | spiceycurry | even thought there are only 3 cables? |
| 21:16.29 | ChannelZ | it configures everything it can see |
| 21:16.34 | spiceycurry | AH ok, great |
| 21:16.37 | spiceycurry | thanks |
| 21:16.52 | ChannelZ | So long as you don't group all 4 span's channels in a group you're trying to dial with, it's fine |
| 21:17.11 | ChannelZ | it'll just make your channel list that much bigger. You can comment out the 4th so no one gets confused |
| 21:17.13 | spiceycurry | Ok, this is much easier than hacking away lol |
| 21:17.23 | cusco | rrb3942: bmoraca_work any hints on why I cannot register? |
| 21:17.49 | ChannelZ | seeing as how this system was dumped on you, might as well try to help the next guy a little.. |
| 21:18.07 | spiceycurry | I totally agree |
| 21:19.35 | rrb3942 | cusco: not sure, did you change the registration line on box2? or the peer information on box1 for box2? |
| 21:20.49 | cusco | I only changed the host=192....blah |
| 21:20.57 | cusco | from dynamic |
| 21:21.10 | cusco | http://paste.debian.net/75898/ |
| 21:21.17 | cusco | SIP/2.0 404 Not Found |
| 21:21.52 | cusco | no authentication on both boxes trying to register on one another |
| 21:22.04 | cusco | Im going backwards, I think I will set the host=dynamic again |
| 21:22.28 | rrb3942 | says wrong password |
| 21:22.34 | rrb3942 | line 69 |
| 21:22.47 | cusco | yes |
| 21:22.54 | cusco | but its exactly the same |
| 21:23.05 | cusco | I double checked |
| 21:23.26 | cusco | :/ |
| 21:23.33 | cusco | I will try a simpler passw |
| 21:24.02 | rrb3942 | that is odd |
| 21:24.38 | cusco | still says wrong apsswd |
| 21:25.26 | cusco | going to change the username... |
| 21:26.03 | *** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br) |
| 21:26.45 | cusco | the same... |
| 21:26.48 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 21:26.53 | cusco | Im running out of ideas |
| 21:27.02 | [TK]D-Fender | cusco: Show something complete |
| 21:27.14 | italorossi | Hello all, can I use a featurecode with only one active channel after Answer() ? |
| 21:27.21 | cusco | [TK]D-Fender: what do you wish to see? |
| 21:27.44 | spiceycurry | I commented out the 4th span, restarted dahdi, but it still shows as RED instead of UNCONFIGURED |
| 21:27.50 | spiceycurry | anything I need to do? |
| 21:27.58 | cusco | spiceycurry: just don't use it to dial |
| 21:28.05 | spiceycurry | ok |
| 21:28.20 | ChannelZ | spiceycurry: who shows red? |
| 21:28.30 | spiceycurry | the 4th span |
| 21:28.37 | spiceycurry | in dahdi_tool |
| 21:28.45 | ChannelZ | no, dahdi_hardware or something? |
| 21:28.47 | ChannelZ | ah |
| 21:28.54 | ChannelZ | yeah it will always see the hardware as a whole |
| 21:29.25 | ChannelZ | what you are configuring is channels available for use in asterisk. |
| 21:29.25 | *** join/#asterisk jamicque (jamicque@jam.bema.one.pl) |
| 21:29.36 | spiceycurry | ok cool |
| 21:29.46 | rrb3942 | cusco: try reverting back before any changes, good luck, time for me to head out |
| 21:29.48 | *** part/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 21:30.14 | italorossi | I'm trying to activate an "secret code" that user can press when hearing an announce.. |
| 21:30.18 | miamiseb | cusco: can I get a sip debug? |
| 21:30.59 | jamicque | hi, does any one knows a way for asterisk to match the correct peer if there are two peers configured with same ip and port. Changin ruri would probably help (but asterisk alone can't change this header). |
| 21:31.08 | ChannelZ | italorossi: the simple way is to just make an extension of 'the secret code' that does something. Then use Background() to play the announcement |
| 21:31.37 | *** join/#asterisk x303 (~X303@wifi222-144.meruwifi.fit.edu) |
| 21:31.40 | *** join/#asterisk sahafeez_ (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 21:31.44 | spiceycurry | I get this error when restarting dahdi - ERROR: Module dahdi_echocan_mg2 is in use (and) ERROR: Module dahdi is in use by dahdi_echocan_mg2,wct4xxp - anything to worry about? |
| 21:31.46 | *** join/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net) |
| 21:32.11 | cusco | spiceycurry: asterisk is using them probably. |
| 21:32.13 | miamiseb | I think it tried to rmmod the kernel modules, which, if they are in use by asterisk, will not be possible. |
| 21:32.25 | cusco | miamiseb: sure hold |
| 21:32.32 | ChannelZ | no-ish, there's a slight problem in how it unloads the drivers and it tries to delete the echo cancelers before the channels using them |
| 21:33.08 | knarfly | I'm trying to setup my dialplan so when I make an outside call I can use *1 to start recording the call, but I'm not sure how to do this? |
| 21:33.11 | ChannelZ | if you stop the drivers twice in a row it shouldn't complain the second time |
| 21:33.23 | spiceycurry | it was asterisk, its all good now |
| 21:34.38 | ChannelZ | ah good. I had the same problem under one of the versions of dahdi, I forget exactly. where it was trying to unload the drivers in the wrong order (or one of them was being slow about stopping |
| 21:34.39 | spiceycurry | now it has restarted, it is no longer RED. it shows unconfigured :D |
| 21:35.33 | italorossi | ChannelZ: Thanks! |
| 21:35.46 | ChannelZ | sure |
| 21:36.23 | knarfly | features show command indicates the *1 feature is active |
| 21:36.45 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.8 (2010/06/01), 1.6.0.28, 1.6.1.20 (2010/05/20), 1.4.32 (2010/06/01), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.1 (2010/06/02) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 21:36.51 | *** join/#asterisk scrooge-mcduck (~scroogey@80-219-24-66.dclient.hispeed.ch) |
| 21:36.52 | scrooge-mcduck | hello |
| 21:37.07 | leifmadsen | libpri 1.4.11.1 has just been released. For more information please see the release announcement at http://www.asterisk.org/node/51364 |
| 21:37.10 | ChannelZ | knarfly: did you specify the 'w' flag or similar in Dial? |
| 21:37.26 | ChannelZ | (probably W) |
| 21:37.44 | miamiseb | you'll need a W in the Dial step so that YOU can start recording a w if you want the receiving party to be able to begin a recording. |
| 21:37.47 | knarfly | Channel Z yes but that is where I'm getting confusrd |
| 21:38.25 | knarfly | exten => _1NXXNXXXXXX,n,Dial(SIP/my_voipprovoder/${EXTEN}) |
| 21:38.39 | scrooge-mcduck | hmm, i am having trouble to issue calls |
| 21:39.21 | knarfly | ChannelZ when I placed the Ww flags in this line it won't work anymore |
| 21:39.28 | spiceycurry | What conditions would I need a trunk group configured? |
| 21:39.35 | ChannelZ | knarfly: core show application dial |
| 21:39.38 | miamiseb | exten => _1NXXNXXXXXX,n,Dial(SIP/my_voipprovoder/${EXTEN},WT) |
| 21:39.42 | ChannelZ | There are multiple arguments. |
| 21:39.53 | ChannelZ | Dial(SIP/blah/${EXTEN},,W) |
| 21:39.59 | miamiseb | or | depending on the version of * |
| 21:40.28 | ChannelZ | miamiseb: need TWO ,'s - the first argument after the device/exten to dial is a timeout. Other args after that. |
| 21:40.30 | miamiseb | yeah, I left out the timeout, so youll want that extra comma ChannelZ threw in |
| 21:40.41 | miamiseb | lol |
| 21:41.02 | cusco | miamiseb: http://paste.debian.net/75901/ |
| 21:41.04 | knarfly | miamiseb: let me try that again but a moment ago it failed |
| 21:41.14 | knarfly | miamiseb: you in Miami too? |
| 21:41.18 | miamiseb | yup |
| 21:41.26 | knarfly | Miami Lakes here! |
| 21:42.00 | cusco | 1.6.2.8 is final |
| 21:42.11 | scrooge-mcduck | grr |
| 21:42.14 | scrooge-mcduck | i keep getting: [Jun 2 23:42:02] NOTICE[31028]: chan_sip.c:20139 handle_request_invite: Call from '101' to extension '1234' rejected because extension not found. |
| 21:42.21 | scrooge-mcduck | anyone know why that could be? |
| 21:42.31 | cusco | there is no dial plan in that context for 1234 |
| 21:42.47 | knarfly | exten => _1NXXNXXXXXX,n,Dial(SIP/my_voipprovider/${EXTEN},wWFotThHr) trying this now! |
| 21:43.12 | [TK]D-Fender | scrooge-mcduck: Because just like it says, the call can't find a match for that number in the context the call is looking in |
| 21:43.39 | cusco | miamiseb: taking a look? |
| 21:43.49 | miamiseb | yup |
| 21:43.52 | ChannelZ | knarfly: it won't work, didn't you listen to me? |
| 21:43.53 | miamiseb | what's pcmedic? |
| 21:44.11 | cusco | miamiseb: the useragent |
| 21:44.12 | miamiseb | knarfly, one extra comma before the flags, so that you let it use the default timeout |
| 21:44.24 | miamiseb | right, but is this a softphone on your machine then or what? |
| 21:44.36 | cusco | no its two asterisk boxes |
| 21:44.44 | cusco | I just felt like setting a useragent for curiosity |
| 21:44.47 | cusco | I can take it out |
| 21:45.24 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:45.26 | miamiseb | Found peer 'gateway' for '932485457' from 192.168.2.7:5060 |
| 21:45.28 | Jumpie | anybody know where you can get SMS short numbers from? like "text love to 31337" |
| 21:45.36 | miamiseb | but you seem to be register a user gateway-sip. |
| 21:45.40 | cusco | it err |
| 21:45.42 | cusco | you're right! |
| 21:45.49 | cusco | I jsut changed the usernames... |
| 21:45.54 | miamiseb | =) |
| 21:46.18 | cusco | why is that "gateway" |
| 21:46.23 | cusco | its set as gateway-sip |
| 21:46.27 | cusco | ?!? |
| 21:46.38 | miamiseb | maybe you didn't reload your sip conf or reload your dialplan after making changes. |
| 21:46.44 | cusco | I did |
| 21:46.46 | cusco | several times |
| 21:47.03 | miamiseb | sip reload and then sip show peers for me |
| 21:47.08 | cusco | ok |
| 21:48.06 | *** part/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 21:48.23 | cusco | http://paste.debian.net/75902/ |
| 21:50.22 | miamiseb | it shows registered. Can you receive/make calls? |
| 21:51.24 | cusco | making call into box1, box2 shows the follwoing sip debug: http://paste.debian.net/75904/ |
| 21:51.31 | scrooge-mcduck | cool |
| 21:51.33 | scrooge-mcduck | stupid me |
| 21:51.38 | scrooge-mcduck | usued wrong context in extensions :) |
| 21:51.47 | cusco | still found peer 'gateway'0 |
| 21:52.06 | cusco | I can change the usernames back.. to primario and gateway... |
| 21:52.43 | cusco | I canno restart the whole asterisk tho, it is live |
| 21:54.21 | cusco | just changed it and sip reloaded twice on each asterisk |
| 21:54.29 | scrooge-mcduck | hmm, how do i configure for incoming calls? |
| 21:54.53 | scrooge-mcduck | aha |
| 21:55.00 | ChannelZ | knowing nothing about your config, we have no idea. |
| 21:55.17 | ChannelZ | thank you, drive through |
| 21:55.57 | cusco | miamiseb: sip debug now: http://paste.debian.net/75906/ |
| 21:56.08 | cusco | Found peer 'gateway' should now be correct |
| 21:56.18 | cusco | tho I there is something stillw rong |
| 21:57.15 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733492.dsl.bell.ca) |
| 21:57.26 | scrooge-mcduck | <PROTECTED> |
| 21:57.50 | ChannelZ | seems clear |
| 21:58.15 | [TK]D-Fender | scrooge-mcduck: Sames as before |
| 21:58.24 | jamicque | does any one knows a way for asterisk to match the correct peer if there are two peers configured with same ip and port. Chaning ruri would probably help (but asterisk alone can't change this header). |
| 21:58.25 | cusco | still [Jun 2 22:57:40] NOTICE[6202]: chan_sip.c:17767 handle_response_invite: Failed to authenticate on INVITE to '"932485457" <sip:932485457@192.168.2.7>;tag=as2515e22a' |
| 21:59.09 | *** join/#asterisk jhirley (~jhirley@adsl-4-130-148.mia.bellsouth.net) |
| 21:59.25 | leifmadsen | jamicque: it needs to match on the [username] part then which I believe is in the From header |
| 21:59.37 | scrooge-mcduck | but i got |
| 21:59.39 | scrooge-mcduck | exten => s,1,Answer :( |
| 21:59.43 | miamiseb | jamicque: how do you get two peers on the same port/ip address pair? Even if its a nat box or something it would hand out different ports. |
| 21:59.51 | miamiseb | im looking cusco |
| 21:59.52 | [TK]D-Fender | scrooge-mcduck: IN THE WRONG PLACE |
| 22:00.02 | ChannelZ | scrooge-mcduck: is it in the same context as whatever channel the call is coming in on is set to? |
| 22:00.12 | [TK]D-Fender | ChannelZ: Obviously not |
| 22:00.13 | cusco | thank you miamiseb |
| 22:00.22 | scrooge-mcduck | uh, how do i find it out then? |
| 22:00.38 | ChannelZ | by reading your configs. We can't. |
| 22:00.43 | [TK]D-Fender | scrooge-mcduck: look at the SIP DEBUG to confirm what peer it's matching and in which context it's looking |
| 22:00.51 | [TK]D-Fender | ChannelZ: No.. not configs. |
| 22:00.59 | [TK]D-Fender | ChannelZ: Doesn't prove anything. The call will. |
| 22:01.03 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 22:01.04 | cusco | miamiseb: peer information: http://paste.debian.net/75892/ |
| 22:01.10 | jamicque | leifmadsen => but those are peers and i change the presentation |
| 22:01.12 | miamiseb | cusco, what are you dialing and from what? what user is the device you are dialing out of trying to register with. And is 932485457 a extension on the other box or a call destined for the outside? If its supposed to be for the US, make sure you enter the extra digit your missing |
| 22:01.32 | leifmadsen | jamicque: if they are only peers, it will only match on IP address |
| 22:01.46 | [TK]D-Fender | And "peers" don't register. |
| 22:01.48 | [TK]D-Fender | CRAZY PEOPLE |
| 22:01.54 | scrooge-mcduck | uh, sip debug? |
| 22:01.58 | leifmadsen | and now I'm heading off to make dinner |
| 22:02.01 | [TK]D-Fender | scrooge-mcduck: CORRECT |
| 22:02.01 | cusco | miamiseb: 932485457 is the DDI from my mobile, box1 has some PRI lines. They all work alright If I Dial trough IAX2 I can reach box2. But I trying sip |
| 22:02.04 | *** part/#asterisk {Repelex} (~{Repelex}@201-71-129-252-arpa.vsp.com.br) |
| 22:02.12 | [TK]D-Fender | leifmadsen: Gotta Be KD! |
| 22:02.16 | ChannelZ | making things complicated.. but ok.. |
| 22:02.20 | scrooge-mcduck | sip debug voipbuster ? |
| 22:02.33 | [TK]D-Fender | scrooge-mcduck: No, full.... not restricted |
| 22:02.40 | [TK]D-Fender | scrooge-mcduck: "sip set debug on |
| 22:03.15 | jamicque | leifmadsen ok, thx |
| 22:03.49 | miamiseb | So what device are you using to dial? a softphone? a voip phone? what user is it registered to your sip server as? Which sip server is it registered to? |
| 22:04.46 | cusco | miamiseb: Im dialing from my mobile phone |
| 22:04.54 | cusco | I am the sip server |
| 22:04.58 | knarfly | Okay, got it...see I told you guys if you'd just listen to ChannelZ it would work 8-) |
| 22:05.11 | [TK]D-Fender | cusco: And how do said magical microwaves REACH ASTERISK? |
| 22:05.19 | cusco | dahdi |
| 22:05.26 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 22:05.40 | [TK]D-Fender | cusco: PASTEBIN <--------- |
| 22:05.47 | cusco | dahdi conf? |
| 22:05.59 | cusco | ok hold |
| 22:06.07 | [TK]D-Fender | cusco: How about all the pieces related to what is broken... along with the failed call to back it up |
| 22:06.16 | citywok | i've had quite a few ghost calls, most of them tend to be in meetme. 30 hours of a call that asterisk thinks is going, but the phone isn't even registered. |
| 22:06.30 | cusco | its only the sip part.. but ok I will paste all of the dial planl, hold |
| 22:06.33 | citywok | i've had the same thing happen, and then whenever i've dialed any phone number from a desk phone it would re-join that conference... |
| 22:06.58 | citywok | if i'm dialing from a phone that has a ghost sip call in meetme that is |
| 22:07.18 | citywok | any ideas what the heck the deal is there? 1.6.1.18 |
| 22:07.45 | scrooge-mcduck | Looking for s in voipbuster-out (domain 192.168.1.3) :D |
| 22:08.18 | miamiseb | cusco: what are you dialing from your mobile phone? A did assigned to the PRI? A did that goes to a ITSP and then gets delivered to you over sip? |
| 22:09.12 | scrooge-mcduck | [Jun 3 00:08:47] WARNING[19927]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) |
| 22:09.16 | scrooge-mcduck | i suppose this aint good? |
| 22:10.16 | paulc | Warm'n'fuzzies: Checking stats on a box and seeing it took 408404 calls in the past 19 weeks.. Nice! :-) |
| 22:11.14 | *** join/#asterisk jhirley (~jhirley@adsl-4-130-148.mia.bellsouth.net) |
| 22:12.10 | [TK]D-Fender | scrooge-mcduck: I would say "no" |
| 22:12.19 | *** part/#asterisk jhirley (~jhirley@adsl-4-130-148.mia.bellsouth.net) |
| 22:12.29 | *** join/#asterisk jhirley (~jhirley@adsl-4-130-148.mia.bellsouth.net) |
| 22:14.04 | cusco | [TK]D-Fender: that is box1 http://paste.debian.net/75913/ |
| 22:14.11 | jhirley | Hey has anyone gotten ADA working with Windowz 7 ? |
| 22:14.21 | scrooge-mcduck | i am not getting it to work, this is the current log: http://pastebin.com/FFzWsfxE |
| 22:14.42 | cusco | miamiseb: ddi is assigned to that pri, asterisk gets the call OK. If the dialplan has a dial trough iax instead, works just fine |
| 22:14.51 | cusco | [TK]D-Fender: forgot the dialplan, hold |
| 22:16.01 | cusco | that is the whole dialplan http://paste.debian.net/75914/ |
| 22:16.06 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 22:16.30 | cusco | miamiseb: I can just try and register a softphone with it instead and use the same kind of dial |
| 22:16.41 | cusco | would you like me to? |
| 22:16.50 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 22:17.09 | miamiseb | I don't see where it receives the call to a DID and tries to find a peer to send it to. The 404 your getting has a from: of pcmedic though, so unless you've got a friend defined as pcmedic, which is what it seems your trying to auth as, it might be problematic. |
| 22:17.20 | [TK]D-Fender | scrooge-mcduck: [Jun 3 00:10:55] WARNING[20933]: chan_sip.c:5328 create_addr: No such host: 1000 |
| 22:17.21 | miamiseb | Can you include the console messages and the SIP debug on a test call? |
| 22:17.26 | [TK]D-Fender | scrooge-mcduck: "sip show peer 1000" |
| 22:17.42 | scrooge-mcduck | Peer 1000 not found. |
| 22:17.44 | scrooge-mcduck | hmm |
| 22:17.48 | scrooge-mcduck | i want to ring all my users :( |
| 22:17.54 | cusco | miamiseb: that is the whole dialplan http://paste.debian.net/75914/ |
| 22:18.00 | [TK]D-Fender | scrooge-mcduck: Well why don't you tell ME why you are dialing a device that doesn't exist... |
| 22:18.07 | cusco | miamiseb: im dialing 210332450 |
| 22:18.13 | miamiseb | Cool, but I'd like to see the console messages as it processes it. |
| 22:18.15 | [TK]D-Fender | scrooge-mcduck: This is your dialplan.. you just pointed it to something that doesn't exist. |
| 22:18.17 | cusco | ok |
| 22:18.18 | scrooge-mcduck | because i wanted to callt his one: exten => 1000,1,Dial(SIP/101&SIP/102,20,tr) |
| 22:18.19 | cusco | hold |
| 22:18.27 | cusco | without sip debug |
| 22:18.28 | cusco | ? |
| 22:18.33 | [TK]D-Fender | scrooge-mcduck: Well that isn't what you told it to do, now is it? |
| 22:18.34 | miamiseb | No with |
| 22:18.39 | grungies1138 | Asterisk Business edition is the same software as switchvox right? |
| 22:18.39 | cusco | ok |
| 22:18.51 | [TK]D-Fender | grungies1138: No |
| 22:19.09 | scrooge-mcduck | i did exten => s,1,Dial(SIP/1000,20,tr) |
| 22:19.21 | [TK]D-Fender | grungies1138: ABE is just a paid for copy of * with a few commercial extras that few people care about, a book, and a bill to make managers feel like they own something |
| 22:19.30 | cusco | miamiseb: http://paste.debian.net/75917/ |
| 22:19.40 | [TK]D-Fender | scrooge-mcduck: Well I guess you should have done what the OTHER exten you showed me would have... |
| 22:19.42 | cusco | that is for box1 only (the one with the PRI lines) |
| 22:19.57 | grungies1138 | lol. gotcha. that and support though right? |
| 22:20.56 | [TK]D-Fender | cusco: Your CLI output isn't showing the failure |
| 22:21.06 | cusco | [TK]D-Fender: yes it is |
| 22:21.08 | [TK]D-Fender | grungies1138: Nope. Thats still hourly |
| 22:21.10 | cusco | line 105 from pastebin |
| 22:21.13 | grungies1138 | hmm. |
| 22:21.18 | [TK]D-Fender | grungies1138: Beyond basic install |
| 22:21.28 | cusco | and downwards |
| 22:21.37 | miamiseb | dialplan from 192.168.2.5 please |
| 22:21.43 | [TK]D-Fender | cusco: which? |
| 22:21.44 | miamiseb | and a sip show peers |
| 22:21.55 | grungies1138 | is the switchbord available somewhere? |
| 22:22.01 | cusco | 23:17 < cusco> miamiseb: that is the whole dialplan http://paste.debian.net/75914/ |
| 22:22.03 | [TK]D-Fender | grungies1138: Which? |
| 22:22.04 | miamiseb | http://paste.debian.net/75917/ |
| 22:22.06 | cusco | its ddi 210332450 |
| 22:22.17 | scrooge-mcduck | http://pastebin.com/4FGBBKi3 |
| 22:22.23 | scrooge-mcduck | i have this in my extensions.conf now |
| 22:22.30 | miamiseb | cusco, the dialplan you gave us was from the .7 right? |
| 22:22.36 | cusco | ah sorry |
| 22:22.38 | russellb | switchvox is ABE + a bunch more stuff |
| 22:22.43 | cusco | do you need dialplan from .5 ? |
| 22:22.43 | [TK]D-Fender | cusco: there is no line 105 in that one |
| 22:22.45 | miamiseb | yes |
| 22:22.51 | cusco | [TK]D-Fender: http://paste.debian.net/75917/ |
| 22:22.54 | grungies1138 | like switchvox's |
| 22:22.58 | cusco | miamiseb: ok hold |
| 22:23.09 | miamiseb | line 105: [Jun 2 23:19:00] NOTICE[6202]: chan_sip.c:17767 handle_response_invite: Failed to authenticate on INVITE to '"932485457" <sip:932485457@192.168.2.7>;tag=as75435f3b' |
| 22:23.12 | [TK]D-Fender | scrooge-mcduck: exten => s,1,Dial(SIP/1000,20,tr) <-doesn't do what you want, so go CHANGE IT |
| 22:23.13 | russellb | grungies1138: the switchvox switchboard is only available with switchvox |
| 22:23.38 | miamiseb | If it's a local extension you don't need the SIP/ |
| 22:23.53 | cusco | miamiseb: http://paste.debian.net/75919/ |
| 22:24.02 | scrooge-mcduck | think its working :P |
| 22:24.05 | cusco | miamiseb: paste line 33 has the 150 |
| 22:24.20 | p3nguin | <miamiseb> If it's a local extension you don't need the SIP/ <--- What? |
| 22:24.25 | cusco | miamiseb: its local on asterisk .5 |
| 22:24.25 | [TK]D-Fender | cusco: Dialplan has nothing to do with auth failures |
| 22:24.39 | cusco | [TK]D-Fender: I thoght it wouldn't |
| 22:24.42 | miamiseb | He's not supposed to be authing |
| 22:24.42 | cusco | thought |
| 22:24.54 | cusco | but the error is regarding an auth failure |
| 22:24.56 | miamiseb | he's dialing a extension that should allow the call to be delivered |
| 22:25.09 | miamiseb | cusco: can you make extension 150 anwser rather than just dial again? |
| 22:25.13 | cusco | yes |
| 22:25.23 | cusco | I just got a call froma colleague (on .5) |
| 22:25.35 | scrooge-mcduck | now it works :) |
| 22:26.15 | miamiseb | does that mean it works? |
| 22:26.20 | cusco | yes |
| 22:26.31 | cusco | I understand that dialplan from .5 is not related, I'm just trying to get help.. |
| 22:26.54 | cusco | the proble is obviously 192.168.2.7 dialing 192.168.2.5 trough sip |
| 22:27.04 | cusco | I can dial trough IAX2 !! |
| 22:27.30 | cusco | but right now I am interested in sip to redirect the audio media |
| 22:28.50 | cusco | and right now host=dynamic in peer |
| 22:29.03 | cusco | if I set it to its host, I cant register due to wrong password |
| 22:29.11 | cusco | even tho password is rightfully spelled! |
| 22:29.29 | cusco | well i guess you folks are tired |
| 22:29.32 | cusco | Im tired too |
| 22:29.40 | cusco | 11pm, time to go home |
| 22:30.01 | miamiseb | cusco: if what you want is the .7 to register to the .5, maybe including a register => line in your sip.conf on the .7 that registers to the .5 |
| 22:30.22 | cusco | miamiseb: I do have that |
| 22:30.52 | miamiseb | p3nguin: I was saying he might try another transport technology if it isn't working with sip the way he wanted. |
| 22:31.01 | miamiseb | cusco: not on the sip.conf you pastebin'd |
| 22:31.10 | cusco | yes it is |
| 22:31.14 | cusco | loog for 192.168.2. |
| 22:31.26 | cusco | look* |
| 22:31.33 | p3nguin | miamiseb: It sounded like you were trying to say you could use Dial(1000) rather than Dial(SIP/1000), which seems silly to me. |
| 22:32.03 | p3nguin | miamiseb: In addition to that, what the hell is a "local extension" anyway? |
| 22:32.21 | miamiseb | One that is defined in your own dialplan, rather than on a remote server |
| 22:32.25 | cusco | p3nguin: he meant a local dialplan |
| 22:32.27 | ChannelZ | one where the plug goes into his mouth |
| 22:32.32 | cusco | where then I could have a sip dial |
| 22:32.36 | cusco | no need to complicate p3nguin |
| 22:33.00 | cusco | well... Im going |
| 22:33.03 | cusco | thanks for trying |
| 22:33.10 | cusco | I might be bugging you again tomorrow |
| 22:37.05 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:37.46 | Defraz | Has anyone had some good luck with Speex? |
| 22:40.25 | grungies1138 | only difference between TDM and AEX Digium cards is PCI vs PCI-E right? |
| 22:41.29 | Qwell | grungies1138: yes |
| 22:41.54 | grungies1138 | sweet thanks |
| 22:43.07 | *** join/#asterisk Beltechs (~Beltechs@netblock-68-183-48-2.dslextreme.com) |
| 22:43.35 | Beltechs | how can I back up my tftp folder to my ftp site automatically |
| 22:43.37 | Beltechs | ? |
| 22:44.06 | grungies1138 | cron? |
| 22:44.55 | Beltechs | but dont i need a sync software? |
| 22:45.05 | Beltechs | im not well versed in linux |
| 22:45.14 | Beltechs | like r sync? |
| 22:45.41 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 22:46.55 | miamiseb | Beltechs: you want it to backup VIA ftp? or just TO an ftp server via whatever technology can get the job done? |
| 22:47.54 | scrooge-mcduck | hmm |
| 22:48.11 | Beltechs | I want to be able to copy/backup my local directory to my ftp when it changes should be automatic |
| 22:48.14 | scrooge-mcduck | what could cause a SIP/2.0 401 Unauthorized on a INVITE/Seq: 684 INVITE for a incoming call? |
| 22:51.31 | [TK]D-Fender | Beltechs: rm + cp |
| 22:52.13 | knarfly | ok so I called ext 201 from ext 101 and made the connx, then 101 transfered the call to ext 700 or parked the call...ext 201 heard moh for about 3 seconds and then the cli filled up with "...Exceptionally long voice lebgth queuing to IAX2/201...." |
| 22:52.21 | miamiseb | But how to do change detection that way? |
| 22:53.09 | miamiseb | Beltechs: you can use rsync, which you mentioned, or you can use the cp command with the update flag on a cron schedule maybe |
| 22:53.27 | miamiseb | other alternatives include subversion or csync2 |
| 22:53.33 | knarfly | when I park a cakk with the IAX2 phone it works fine but when the SIP phone parks the call....zoom and the log file fills up |
| 22:53.56 | knarfly | only way out is Ctrl+C |
| 23:02.45 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 23:03.01 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 23:03.54 | Defraz | I can't get the asterisk source to pick up on the speex codec library |
| 23:03.59 | *** join/#asterisk sahafeez_ (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 23:04.15 | Defraz | it says in all the howto's that it will pick the lib up if it is installed when recompiling asterisk. |
| 23:04.20 | Defraz | Any ideas? |
| 23:04.28 | Defraz | I have checked the ld.config file. |
| 23:04.53 | scrooge-mcduck | damn friggin sipcall |
| 23:10.04 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:12.16 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 23:16.00 | *** join/#asterisk schmick1 (~schmick1@60-242-87-233.static.tpgi.com.au) |
| 23:20.47 | scrooge-mcduck | aweet finally wokring |
| 23:20.49 | scrooge-mcduck | sleep well all! |
| 23:22.00 | ChannelZ | No! Mom said I could stay up another 2 hours. |
| 23:22.45 | grungies1138 | Get to bed now, or no Xbox tomorrow! |
| 23:40.10 | vader-- | to allow an outside SIP phone to connect to an asterisk box I only need port UDP/5060 and UDP/10000-20000 open to the asterisk box? |
| 23:43.03 | knarfly | vader--: that should do it if the SIP phone is connecting diretcly to you * server |
| 23:44.10 | knarfly | so is your * server hanging right off your internet connex |
| 23:44.31 | vader-- | asterisk <--> router/firewall <--> Internet |
| 23:45.10 | knarfly | Good...you will need to do port forwarding as well from your router to yout * box |
| 23:51.14 | grungies1138 | are the AA50s worth it? |
| 23:54.29 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |