| 00:07.37 | ziz212 | [TK]D-Fender, lorsungcu now it is working because of your comments .. thanks |
| 00:21.06 | john_____ | Hi |
| 00:21.19 | john_____ | can anyone help me with this thread in digium forum? http://forums.digium.com/viewtopic.php?t=73703 |
| 00:21.28 | john_____ | it is a known issue since 1.6.x days. |
| 00:22.19 | [TK]D-Fender | its in the sip.conf sample.... |
| 00:22.58 | john_____ | im currently using Asterisk 1.8 |
| 00:23.10 | carrar | INCORRECRT RESPONSE |
| 00:23.25 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 00:23.29 | [TK]D-Fender | I love turtles. |
| 00:23.36 | carrar | heh |
| 00:24.06 | carrar | esculating that 4xx to 6xx |
| 00:24.14 | carrar | heh |
| 00:27.42 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 00:28.39 | john_____ | [TK]D-Fender: sip.conf sample? |
| 00:29.07 | carrar | that came with the source |
| 00:29.12 | carrar | that you installed |
| 00:29.29 | john_____ | im reading it at http://www.digium.com/asterisk_handbook/sip.conf.pdf |
| 00:29.40 | john_____ | and /etc/asterisk/sip.conf |
| 00:29.53 | john_____ | am i really forced to use back Asterisk 1.4? :| |
| 00:30.36 | john_____ | my username contains one of the reserved characters "@" |
| 00:35.14 | carrar | john_____, how many SIP devices do you have with "@" in the anme? |
| 00:37.52 | *** join/#asterisk tamiel (~tamiel@64-71-8-110.static.wiline.com) |
| 00:40.57 | john_____ | carrar: just one with my provider. |
| 00:41.13 | carrar | can they cahnge that? |
| 00:41.21 | carrar | or you change your user name |
| 00:41.30 | john_____ | impossible. as they rolled it out with 200,000 nation wide users |
| 00:41.31 | carrar | to one without a @ |
| 00:41.40 | carrar | thats insaine |
| 00:41.47 | carrar | what carrier is that |
| 00:41.56 | john_____ | the numbers have outgrown what i mentioned |
| 00:41.59 | john_____ | that was 2 years back |
| 00:42.04 | john_____ | www.m1.com.sg |
| 00:42.17 | john_____ | we are forced to comply to the business voip service with their gateway |
| 00:42.28 | john_____ | and it sucks |
| 00:44.00 | carrar | so you just need @ in the register line |
| 00:44.04 | carrar | for the username |
| 00:44.26 | john_____ | thats right carrar! |
| 00:44.49 | john_____ | i have it right there in the register line. |
| 00:44.53 | john_____ | WIMPy helped me with it. |
| 00:45.07 | john_____ | but Asterisk seems to take it out during REGISTRATION |
| 00:45.11 | carrar | authuser=123400@xyz.com |
| 00:45.13 | john_____ | SIP registration |
| 00:45.32 | carrar | point your reg line to a contect |
| 00:45.37 | carrar | context |
| 00:45.46 | john_____ | ah. how do i do that? |
| 00:45.48 | carrar | show me what you have reg line and context |
| 00:45.52 | carrar | from your sip.conf |
| 00:45.55 | john_____ | i have authuser=123400@xyz.com as you mentione. |
| 00:46.02 | john_____ | so, |
| 00:46.05 | john_____ | register => authuser ? |
| 00:46.10 | carrar | show me the butter |
| 00:46.17 | carrar | binpaste |
| 00:46.21 | carrar | ~binpaste |
| 00:46.21 | ziz212 | Hi, all how can we get audible CLI to extension? |
| 00:46.28 | john_____ | ok one sec. |
| 00:49.21 | carrar | looks out the window |
| 00:49.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 00:51.02 | carrar | so anyways |
| 00:51.09 | carrar | at the end of your register line |
| 00:51.12 | carrar | <PROTECTED> |
| 00:51.22 | carrar | XXXXXX points to your context with the authuser line |
| 00:51.34 | [TK]D-Fender | ... pardon? |
| 00:51.38 | carrar | well |
| 00:51.44 | carrar | phear my wording |
| 00:51.55 | [TK]D-Fender | Oh I'm afraid.... |
| 00:51.56 | carrar | was wating for a paste |
| 00:51.59 | carrar | so I could just do it |
| 00:53.15 | carrar | peer |
| 00:54.04 | john_____ | ah |
| 00:54.05 | john_____ | ok! |
| 00:54.07 | john_____ | wait |
| 00:54.21 | john_____ | http://pastebin.com/emgMAhGh |
| 00:54.37 | john_____ | i was VPNing through multiple segments |
| 00:56.31 | john_____ | let me do what you suggested |
| 00:56.49 | carrar | put your register line at the top |
| 01:00.19 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-193.mobile.uci.edu) |
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| 01:10.37 | carrar | & |
| 01:18.06 | [TK]D-Fender | exorcises carrar |
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| 02:01.47 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 02:06.48 | john_____ | carrar: i just fucked elastix |
| 02:06.58 | john_____ | im going back to command-line asterisk |
| 02:07.04 | john_____ | do you think i should use asteriskNow or Asterisk? |
| 02:07.48 | john_____ | anyone has suggestions? |
| 02:13.47 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 02:15.08 | jpsharp | Command line asterisk. |
| 02:15.25 | jpsharp | Just plain asterisk. |
| 02:22.04 | *** part/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
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| 02:27.59 | mazpe | what can i be doing wrong now that i'm getting one way voice? |
| 02:28.05 | mazpe | http://pastebin.com/sNHkFvpe |
| 02:28.29 | mazpe | more natting nightmare |
| 02:29.47 | jpsharp | One way which way? |
| 02:29.58 | john_____ | jpsharp: great thanks |
| 02:34.17 | jpsharp | But I've been monkeying with asterisk for 10 year, so I'm biased :) |
| 02:34.40 | mazpe | jpsharp: I can hear the voice in sipphone.. but i hear nothing on my cellphone. |
| 02:37.03 | jpsharp | Do you have the appropriate RTP ports forwarded on the firewall? |
| 02:38.14 | mazpe | Well my asterisk is behind a firewall... RTP 10000:10100 and they are forward to the pbx |
| 02:38.32 | mazpe | my sipclient is in a remote office |
| 02:42.39 | jpsharp | Well, the SIP invites look right, setting the RTP destination to be the firewall's external IP. |
| 02:42.58 | elguero | mazpe: What has ip [10.1.1.11]? |
| 02:43.35 | jpsharp | That's one of the interfaces on the Asterisk box. |
| 02:44.10 | elguero | That is what it looked like... so, this box has three interfaces in total? |
| 02:45.56 | mazpe | elguero: thats the firewall internal IP |
| 02:46.26 | mazpe | the pbx has two internfaces... one 10.2.1.151 which is how it communicates witht he internal network |
| 02:46.52 | mazpe | and the second 192.168.2.253 which is seperate vlan for the polycom phones. |
| 02:47.14 | ChannelZ | We're not seeing the traffic to/from PSTN1 |
| 02:47.32 | mazpe | the firewall has two ifs as well. one with the external ip address and one with the internal |
| 02:47.33 | ChannelZ | Have you also done a simple Echo test on your softphone? Between it and Asterisk, do you get bi-directional audio? |
| 02:47.44 | mazpe | echo test |
| 02:48.04 | ChannelZ | yes.. make an extension that calls Echo() and dial it |
| 02:49.05 | ChannelZ | the problem with all this is that you have 2 calls.. one between your softphone and Asterisk, and one between Asterisk and your ITSP. Figure out which one is broken first. |
| 02:50.11 | mazpe | ChannelZ: well the PSTN1 is local to the pbx |
| 02:50.17 | mazpe | in the same network. |
| 02:50.37 | ChannelZ | I thought that was your 'cell phone' |
| 02:51.06 | mazpe | PSTN1 is a Patton FXO Gateway |
| 02:51.12 | mazpe | i'm using that to dial out |
| 02:51.26 | mazpe | the number been dialed is my cell phone. |
| 02:51.40 | ChannelZ | Oh. I don't remember that having ever been mentioned |
| 02:52.10 | mazpe | :( sorry |
| 02:52.50 | ChannelZ | So you have audio coming from the FXO->Softphone (from the cell) but no audio making it from the Softphone->FXO? |
| 02:54.38 | mazpe | sounds about right.. Because i can hear my hello on the softphone.. but i do not get anything on my cell phone. |
| 02:54.50 | ChannelZ | And the results of the Echo test? |
| 02:58.32 | mazpe | ChannelZ: what is echo suppose to do? because i dont see much |
| 02:58.58 | mazpe | exten => 777,1,Echo |
| 02:59.12 | ChannelZ | It's not what you see, it's what you hear. |
| 02:59.17 | ChannelZ | You speak, it echos back. |
| 02:59.53 | mazpe | nada |
| 03:00.01 | mazpe | unless i'm doing it wrong. |
| 03:00.04 | jpsharp | points fingers at ye olde firewall. |
| 03:00.08 | ChannelZ | It's also helpful to put a Playback(hello) or something prior to Echo, so you can see which way is broke (if not both) |
| 03:00.19 | mazpe | ok |
| 03:00.46 | ChannelZ | So, if you hear 'Hello' and then get jabber away but don't hear yourself back, you know audio isn't making it from the device to Asterisk. |
| 03:00.57 | ChannelZ | s/get // |
| 03:01.38 | mazpe | testing |
| 03:02.32 | SeRi | my flukso is on the way :) |
| 03:03.01 | mazpe | ChannelZ: ok... now that worked. |
| 03:03.15 | mazpe | I got a hello.. and echo hellllllooo hellllllo lol |
| 03:03.18 | ChannelZ | You heard the hello, and your own audio echoing back? |
| 03:03.31 | mazpe | yup |
| 03:03.58 | ChannelZ | ok. So have you tested your FXO gateway singally too? IE dial into your number from your cell and do a similar test? |
| 03:04.50 | ChannelZ | s/singally/singly/ |
| 03:04.53 | ChannelZ | I like to make new words. |
| 03:06.15 | mazpe | so like answer the did with playback(hello) and then echo |
| 03:06.36 | ChannelZ | yeah |
| 03:09.16 | mazpe | ChannelZ: the echo is not perfect... but i can sort of hear it. |
| 03:09.35 | ChannelZ | uhmmm.... |
| 03:09.57 | mazpe | ChannelZ: i can give you a number to dial.. see what you think. |
| 03:10.15 | ChannelZ | it might be your thing getting confused and doing long echo cancellation.. try this instead |
| 03:10.31 | jpsharp | Probably hearing some sort of crosstalk. |
| 03:10.45 | ChannelZ | Instead of Echo, do a Record(testfile.ulaw,,10) followed by Playback(testfile) |
| 03:11.39 | ChannelZ | if you're not getting solid audio then you need to start digging there first |
| 03:11.54 | ChannelZ | (or maybe your cell has terrible reception?) |
| 03:12.30 | mazpe | testing |
| 03:13.16 | mazpe | sounds clean |
| 03:13.41 | ChannelZ | ok so that's all that was probably. |
| 03:13.47 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 03:14.01 | lorsungcu | anyone use swift()/cepstral? |
| 03:14.53 | ChannelZ | So last test. For the extension that is doing the dial out to SIP/PSTN when you dial your cell; For fun make the first thing it does a Playback(hello) and see if it magically works |
| 03:15.05 | mazpe | ChannelZ: the system currently receives/transmits calls via the PSTN1 to local phones with no problem.. |
| 03:15.17 | mazpe | ok |
| 03:17.14 | *** join/#asterisk iq (~iq@ip24-252-27-163.om.om.cox.net) |
| 03:21.18 | mazpe | uhm. |
| 03:21.22 | mazpe | i think that worked. |
| 03:21.35 | lorsungcu | uhm. |
| 03:22.02 | ChannelZ | damn. |
| 03:22.12 | mazpe | wtf? |
| 03:23.10 | ChannelZ | hard to explain. |
| 03:23.16 | mazpe | ok.....now i'm even more confused |
| 03:24.01 | mazpe | i am port forwarding in the remote office 5060, 10000:10100 to the softphone.. which is not a viable solution anyways, because i need to put like 4 more phones here. |
| 03:24.14 | *** join/#asterisk jakent (~jakent@2001:470:8:1fc:b123:e6ee:66e7:1078) |
| 03:24.19 | ChannelZ | I'd have to do some testing to remember but I ran into some oddities with NAT and Asterisk a couple of months ago. |
| 03:24.22 | mazpe | but doing the Playback first opens the audio bidirectional? |
| 03:25.17 | ChannelZ | Well the firewall on the softphone side just needs to allow all traffic going OUT, and ideally to make things easy port-forward audio IN on whatever port your softphone is set to (can be configured in Zoiper) |
| 03:25.26 | mazpe | i comment the playback and it stops working. |
| 03:25.27 | mazpe | lol |
| 03:26.03 | mazpe | how about STUN can it be the answer to all of this? |
| 03:27.39 | ChannelZ | well I think STUN only really deals with external IP discovery here. |
| 03:28.34 | ChannelZ | For the heck of it have you tried port forwarding on the softphone's firewall the port listed in Zoiper's Advanced/rtp options? |
| 03:29.14 | mazpe | it was 8000 before.. i switched it to 10000 |
| 03:32.23 | ChannelZ | which is forwarded to the machine zoiper is running on? |
| 03:33.13 | mazpe | yup |
| 03:34.13 | ChannelZ | hmm |
| 03:35.03 | ChannelZ | and similarly on the firewall Asterisk is behind right? |
| 03:35.30 | mazpe | yeah |
| 03:36.14 | ChannelZ | hmm well that's a puzzle then, I've never seen this behavior when everything is forwarded correctly |
| 03:37.18 | ChannelZ | if you do an 'rtp set debug on' and do a call (with it setup in the 'normally failing' mode with no Playback) does it look like * is receving RTP and sending RTP to all the correct IPs? |
| 03:37.37 | ChannelZ | (warning this will flood your console a bit) |
| 03:40.14 | mazpe | hmm... i just noticed that the STUN was enabled |
| 03:40.24 | mazpe | on zoiper. |
| 03:40.32 | mazpe | i disabled it.. and it seems to be working? |
| 03:42.01 | mazpe | yup. |
| 03:42.06 | mazpe | check this out! |
| 03:42.10 | mazpe | loving wireshark |
| 03:42.18 | mazpe | I made the call running wireshark.. |
| 03:42.35 | mazpe | and i notice this weird address (which was the stun server in zoiper) |
| 03:43.07 | mazpe | so... even with the stun on if i do playback before making the call.. the audio works |
| 03:43.27 | mazpe | disabling the stun makes it work |
| 03:43.38 | mazpe | with out the playbback |
| 03:46.42 | ChannelZ | hmm |
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| 03:51.13 | mazpe | yup.. i just made a call to American Airlines. |
| 03:51.36 | mazpe | with the ports on the firewall from the remote offfice off.. |
| 04:04.05 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 04:05.40 | mazpe | btw, do i have to define dtmf in each of the sip clients? or just on the trunk of the provider? |
| 04:05.48 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 04:08.27 | jpsharp | mazpe: You can do either. |
| 04:10.26 | mazpe | currently is not working been set on the trunk.. testing setting on the client. |
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| 05:13.48 | RiceCracker | just upgraded to asterisk 1.8 from 1.2 now when i make calls to outside lines, i get two different ring back at the same time |
| 05:14.00 | RiceCracker | anyone know of this type of issue? |
| 05:14.59 | kaldemar | what do you mean by "outside lines" specifically? what does your dialing extension do? |
| 05:15.02 | kaldemar | ~pb |
| 05:15.03 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
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| 05:48.24 | RiceCracker | kaldemar, when you make an outbound call you get two types of ring playbacks. you get a single ring, then it doubles, then single/double ring back...when the call connects no issue, but the ring pattern is annyoing to the clients |
| 05:49.54 | RiceCracker | i read something about 180 and 182 ringing seetings, but the information was not too clear on excatly what should be set. |
| 05:50.29 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 05:53.10 | kaldemar | RiceCracker: i got that. now what's your outbound connection and what exactly does your extension do when it dials out? |
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| 05:58.44 | RiceCracker | outbound is over a t1 trunk. What i setup was a short dial key for mobile phones. example: i created a custom extension called 2006. this extension is set to dial a mobile number by going though the t1. when i call the mobile phone normally it rings normal. when i call though the short dial i get the extra ring with the normal ring |
| 06:02.51 | kaldemar | and what does the extension look like? |
| 06:03.33 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:03.36 | schmidts | good morning |
| 06:05.15 | RiceCracker | extension is set to dial DAHDI/g0/98801294 |
| 06:05.29 | kaldemar | RiceCracker: show the whole extension. |
| 06:05.55 | kaldemar | or CLI output of a whole call, rather. |
| 06:06.41 | RiceCracker | hold on, let me make a call and do the cli output |
| 06:08.04 | RiceCracker | <PROTECTED> |
| 06:08.04 | RiceCracker | <PROTECTED> |
| 06:08.05 | RiceCracker | <PROTECTED> |
| 06:08.05 | RiceCracker | <PROTECTED> |
| 06:08.05 | RiceCracker | <PROTECTED> |
| 06:09.07 | RiceCracker | that is with the short key.. |
| 06:09.14 | RiceCracker | this is without |
| 06:09.26 | RiceCracker | -- Called DAHDI/g0/98801294 |
| 06:09.26 | RiceCracker | <PROTECTED> |
| 06:09.26 | RiceCracker | <PROTECTED> |
| 06:09.26 | RiceCracker | <PROTECTED> |
| 06:10.17 | Dar1us | RiceCracker: use pastebin, yeesh |
| 06:10.28 | RiceCracker | sorry |
| 06:11.40 | RiceCracker | kaldemar lets pm on this, i'll brb |
| 06:11.45 | kaldemar | and the _whole_ output with verbosity enabled (core set verbose 10). |
| 06:11.56 | kaldemar | RiceCracker: let's keep it on the channel. |
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| 07:07.15 | ChannelZ | does a one-man pants-off dance-off |
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| 08:28.57 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 08:29.25 | as001 | Hello is there a limit how many agents asterisk manager interface can handle ? |
| 08:30.11 | as001 | I increased number of agents from 80 to 160 i have no load or swap on server but AMI got write error 3 |
| 08:30.56 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 08:40.09 | as001 | Does it have something with writetimeout in manager.conf. It looks like AMI can not write info and has timeout and then perl script which is connected to AMI reconnects because of timeout. I will try to increase writetimeout to extremly high value... |
| 08:40.58 | as001 | My question is Is there any limit how many agents Asterisk Manager Interface can handle ? |
| 08:41.23 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:44.54 | kaldemar | as001: not in software. |
| 08:46.20 | as001 | so do you think I can have 160 agents and I will be able to read/write from/to AMi without Read and Write errors because of timeout in manager.conf ? |
| 08:46.48 | Dar1us | I doubt the timeout is the problem |
| 08:46.58 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 08:48.35 | Dar1us | as001: do your agents maintain a persistent connection to asterisk, or open a connection, issue some commands and exit? |
| 08:49.03 | as001 | Persistent connection have my perl script which comunicate with ami and issue some commands and read some events from ami |
| 08:49.31 | Dar1us | hm I suppose the default timeout is only 100msec |
| 08:49.43 | Dar1us | you could try making it 1sec and see what happens |
| 08:50.05 | as001 | ok I will try. Thanks |
| 08:51.11 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 08:51.18 | *** part/#asterisk as001 (~uros@82.117.198.142) |
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| 09:56.02 | *** join/#asterisk casix (~casix@114.155.23.95.dynamic.jazztel.es) |
| 09:56.04 | casix | hello |
| 10:00.22 | casix | i have upgraded and asterisk to certified 1.8.11-cert5, dahdi-complete-2.6.1, and libpri1.4.12 with an TE200p card. I have compiled all, installed all, the dahdi detects the TE200 card ok but there is no such pri cmd and the calls from or to the pri card are not working. There have been any change in the 1.8 with the dahdi/pri module? |
| 10:01.22 | kaldemar | casix: sounds like you don't have chan_dahdi.so loaded. what does "module load chan_dahdi.so" tell you? |
| 10:02.53 | casix | yes its working, if I execute the load command: "WARNING[17371]: loader.c:840 load_resource: Module 'chan_dahdi.so' already exists." If execute the reload command: "-- Reloading module 'chan_dahdi.so' (DAHDI Telephony Driver)" |
| 10:02.55 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:04.42 | casix | the "dahdi show status" recognize the card correctly with no alarms |
| 10:06.17 | kaldemar | obviously it's not working if you don't have the pri commands in CLI. |
| 10:06.34 | kaldemar | did you compile and install libpri before dahdi and dahdi before asterisk? |
| 10:06.41 | casix | but there is no such pri cmd. I have the pri libs installed "/usr/lib/libpri.a /usr/lib/libpri.so /usr/lib/libpri.so.1.4" |
| 10:07.00 | casix | mmm i'm not shure |
| 10:07.09 | casix | I'm going to recompile dahdi and then asterisk |
| 10:09.13 | kaldemar | "but there is no such pri cmd" <-- elaborate on this. what command exactly? |
| 10:10.41 | casix | "No such command 'pri' (type 'core show help pri' for other possible commands)" |
| 10:11.24 | casix | maybe in 1.8 there is no pri command but old versions had it. |
| 10:13.28 | kaldemar | "pri" alone is not a command. |
| 10:14.17 | casix | but it was... |
| 10:15.18 | casix | "CLI> pri show version" -> "libpri version: 1.4.7" |
| 10:16.26 | WIMPy | So pri is there. What's missing? |
| 10:16.38 | casix | no no, this is from another computer |
| 10:16.57 | WIMPy | Well, then go recompile. |
| 10:17.01 | casix | with old 1.4.36 asterisk |
| 10:17.06 | casix | i'm recompiling |
| 10:17.14 | WIMPy | Order does matter. |
| 10:17.24 | casix | ok |
| 10:17.25 | casix | :) |
| 10:17.28 | tzafrir_laptop | Interesting. http://wiki.openvox.cn/index.php/Enhanced_CID_Detection_Feature_in_A810/1610/2410 |
| 10:18.08 | tzafrir_laptop | I took a look at the drivers. From what I see they do the DSP in the host CPU (but in the driver) |
| 10:19.46 | WIMPy | For Asterisk 1.2? |
| 10:22.50 | casix | WIMPy, kaldemar: yes, order does matter, I have the command, thanks :) |
| 10:23.03 | *** join/#asterisk mintos (~mvaliyav@117.206.23.163) |
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| 11:59.05 | *** join/#asterisk kaspar (~kaspar@p50811EF2.dip.t-dialin.net) |
| 12:00.22 | kaspar | hey, how do I get rtp jitter values in the dialplan? RTPAUDIOQOSJITTER(BRIDGED) seems to be either empty or all zero, depending on who hangs up and wether I call it after Dial(SIP/*) or in h,n,.... |
| 12:00.56 | kaspar | (I want to get the values from an outgoing SIP connection) |
| 12:01.43 | mazpe | morning |
| 12:03.06 | kaspar | morning ;) |
| 12:05.55 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:19.46 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 12:22.26 | *** join/#asterisk tully` (Tully@74.197.171.224) |
| 12:23.26 | tully` | Is there any way to catch congestion or forbidden errors in the dialplan? The only ones I see are timeouts, hangups, and invalid. Neither of these seem to be working. |
| 12:25.07 | WIMPy | ${HANGUPCAUSE} |
| 12:25.19 | leifmadsen | or ${DIALSTATUS} depending on what you're doing |
| 12:26.25 | tully` | I'm just trying to do periodic checks on a number to ensure uptime. I'm guessing the best way is just with a .call file. |
| 12:26.37 | leifmadsen | well, the best way is via the AMI |
| 12:26.48 | leifmadsen | but a .call file would probably work |
| 12:28.50 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-18.nwrknj.fios.verizon.net) |
| 12:29.02 | tully` | awesome, thanks guys I've got a good starting point now |
| 12:41.18 | v0lZy | anyone have a dell powerconnect switch? |
| 12:42.02 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:43.14 | *** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 12:47.30 | [TK]D-Fender | v0lZy, Next non-meta question please.... |
| 12:59.31 | v0lZy | sorry, just trying to ask everywhere if anyone has them |
| 12:59.46 | v0lZy | im unfamiliar with the interface.... and i cant find what im looking for online |
| 13:00.21 | WIMPy | No question, no answer. |
| 13:00.34 | *** join/#asterisk serafie (~erin@nat/digium/x-ifysvyyjmqwpcsqm) |
| 13:00.37 | [TK]D-Fender | And we still don't know what you're looking for. |
| 13:01.28 | *** part/#asterisk mirela666 (~mirko@212.200.146.253) |
| 13:02.43 | v0lZy | If anyone has a dell powerconnect 5500 series switch, im looking how to reset it to factory settings. |
| 13:03.39 | Roelt | with the serial port? |
| 13:04.52 | v0lZy | yeah |
| 13:05.04 | v0lZy | I have a serial connection to it, for some reason i cant get the https or http working |
| 13:05.15 | v0lZy | (which is probably my fault after changing the ip) |
| 13:05.25 | v0lZy | but even then its nowhere in the web interface |
| 13:05.32 | v0lZy | revert to factory settings or something |
| 13:05.43 | Roelt | "ip http server" looks a command to enable the webserver |
| 13:05.49 | *** join/#asterisk mintos (~mvaliyav@117.206.29.79) |
| 13:06.02 | v0lZy | yeah |
| 13:06.04 | v0lZy | tried it |
| 13:06.06 | v0lZy | doesnt wor |
| 13:06.08 | v0lZy | k |
| 13:06.12 | v0lZy | so .. trying to reset to factory defaults |
| 13:07.16 | Roelt | afaik, reload does that.. |
| 13:07.39 | Roelt | command structure looks a lot like the older cisco switches i used to work with |
| 13:09.35 | v0lZy | that appears to just reset the switch |
| 13:09.41 | v0lZy | but not to factory defaults |
| 13:12.00 | v0lZy | i think the reason im not able to access the web interface might be a route problem |
| 13:12.02 | v0lZy | since i changed the ip |
| 13:12.12 | Roelt | hmm.. you can load a configuration |
| 13:12.33 | Roelt | and the device seemd to have a startup-config in the directory, it's possible that's the default config |
| 13:13.10 | Roelt | hmm, no, seems to be the config the unit booted with.. |
| 13:15.09 | v0lZy | friggin frustrating. |
| 13:15.21 | v0lZy | is a factory reset basically flashing the firmware? |
| 13:15.24 | v0lZy | in this case |
| 13:15.36 | Roelt | looks like it |
| 13:15.37 | *** join/#asterisk drhonk (~drhonk@static.57.112.4.46.clients.your-server.de) |
| 13:16.10 | v0lZy | delete startup-config |
| 13:16.40 | v0lZy | hm... dangerous stuff. |
| 13:16.59 | WIMPy | Why? |
| 13:18.04 | v0lZy | yeah |
| 13:18.06 | v0lZy | that did it |
| 13:18.38 | Roelt | ah, nice. |
| 13:18.41 | v0lZy | well since its not documented |
| 13:18.45 | Roelt | sounds scary |
| 13:19.18 | v0lZy | ok, now this appears to be working |
| 13:21.29 | v0lZy | hm |
| 13:21.34 | v0lZy | ok.. how to set time now... lol |
| 13:21.39 | v0lZy | this is like school all over again |
| 13:21.55 | Roelt | haha |
| 13:22.10 | Roelt | clock set hh:mm:ss monyh day year |
| 13:22.29 | v0lZy | i set it to sntp |
| 13:22.34 | Roelt | yeah, that's better :) |
| 13:22.38 | WIMPy | ntp? |
| 13:22.40 | v0lZy | yeah |
| 13:22.48 | v0lZy | but i dont know how to force reload |
| 13:22.52 | v0lZy | of the time |
| 13:22.56 | v0lZy | it still thinks its october 2006 :D |
| 13:23.35 | thecardsmith | Next thing you know it's gonna sign onto myspace and make a post on orkut |
| 13:24.16 | v0lZy | lol :D |
| 13:25.10 | v0lZy | has a poll interval of 1024 seconds |
| 13:25.21 | v0lZy | lets see if it updates anything if i set that to 60 |
| 13:26.18 | v0lZy | and enabled poll unicast.. |
| 13:26.52 | v0lZy | show clock |
| 13:26.54 | v0lZy | didnt know that one :D |
| 13:28.15 | v0lZy | well it still thinks its october 2006... |
| 13:28.40 | *** join/#asterisk _val_ (fbsd@178-84-212-245.dynamic.upc.nl) |
| 13:29.00 | v0lZy | ah, ok, fixed |
| 13:29.18 | _val_ | Hi there. It's me again yes :>. I'm getting this error when I put these two lines. but first the error: [Aug 22 15:26:53] WARNING[2397]: pbx.c:7465 add_pri_lockopt: Unable to register extension '_X.', priority 1 in 'incoming', already in use |
| 13:30.04 | *** part/#asterisk WARlrus (~matt@puma-mxisp.mxtelecom.com) |
| 13:30.17 | _val_ | 2 |
| 13:30.18 | _val_ | exten => _X.,1,ExecIf($["${CALLERID(num):-9}" = "0616855646"]?Hangup(21)) |
| 13:30.18 | _val_ | exten => _X.,1,ExecIf($["${CALLERID(num):-9}" = "0612486548"]?Hangup(21)) |
| 13:30.31 | WIMPy | The priorities need to be numbered in sequence. |
| 13:30.36 | _val_ | So I see it's the prioritie.s |
| 13:30.38 | v0lZy | because u have it twice |
| 13:30.41 | v0lZy | change the bottom one to n |
| 13:30.45 | _val_ | oh I see. |
| 13:30.45 | WIMPy | Use n to get it automatically. |
| 13:30.49 | v0lZy | but |
| 13:31.22 | v0lZy | i think you're gonan do something weird that way :D |
| 13:31.38 | _val_ | WIMPy: you mean exten => _X.,n,ExecIf($["${CALLERID(num):-9}" = "0612486548"]?Hangup(21)) |
| 13:31.41 | _val_ | ? |
| 13:31.43 | _val_ | but can this n be 2? |
| 13:31.48 | WIMPy | yes |
| 13:32.04 | WIMPy | You can number them manually, sure. |
| 13:32.05 | v0lZy | makes more sense to use n so u dont have to re-number all of them when u change something |
| 13:32.33 | _val_ | but is 'n' a valid parameter if you may call it so? |
| 13:32.35 | WIMPy | Indeed. |
| 13:32.54 | _val_ | the above line with the 'n' in it is totally valid? |
| 13:32.57 | WIMPy | And yu are aware, that your _X. will only match if nothing more specific matches? |
| 13:33.58 | _val_ | Within the company we've different numbers. So it blocks the incoming call on all numbers right? |
| 13:34.05 | *** join/#asterisk navaismo (~navaismo@189.144.212.238) |
| 13:34.19 | WIMPy | Only if all calls go to that extension. |
| 13:35.21 | _val_ | To what extension. The intention is this. Block 0612486548 to all incoming extensions |
| 13:36.23 | WIMPy | YOu have to do it on all extensions then. |
| 13:36.57 | WIMPy | Or you send them to another context that does the filtering and then calls your current context afterwards. |
| 13:37.47 | v0lZy | neat, POE works |
| 13:39.58 | _val_ | Hmm.. this is not my cup of coffee >:< |
| 13:40.29 | *** join/#asterisk vizzy (~volker@p54B1A5FA.dip.t-dialin.net) |
| 13:41.10 | leifmadsen | _val_: moar whiskey! |
| 13:41.37 | _val_ | does not drink. |
| 13:41.45 | vizzy | hi :) i dont get it. i try to make a ivr menu to set a pincode which should be stored in astdb, but i dont really understand the syntax of the authenticate() function when using the astdb. |
| 13:41.51 | _val_ | It will kill all my remaining brain cells |
| 13:42.14 | vizzy | so as i understand not the value is used but the key |
| 13:42.22 | vizzy | which sound weird to me |
| 13:42.42 | vizzy | does that mean i can only store one pin in a dbfamily entry? |
| 13:42.47 | leifmadsen | vizzy: http://ofps.oreilly.com/titles/9781449303822/DialplanFundamentals.html |
| 13:42.53 | leifmadsen | vizzy: documented in one of our recipes |
| 13:43.03 | vizzy | leifmadsen, thx i already read through that stuff |
| 13:43.13 | vizzy | maybe i should read again, wait :) |
| 13:43.21 | leifmadsen | wonders what the problem is then |
| 13:43.29 | leifmadsen | back to hacking on EC2! |
| 13:43.48 | vizzy | e.g. i have exten => _1,n,Set(CURPIN=${DB(int_conference_passwd/room551)}) |
| 13:43.48 | vizzy | exten => _1,n,sendText("PIN FOR ROOM 501 IS ${CURPIN}") |
| 13:43.59 | *** join/#asterisk DynamicFail (~tmccurdy@147.177.61.207) |
| 13:44.14 | vizzy | ^ to read the pin out of the db |
| 13:44.28 | leifmadsen | why _1? |
| 13:44.32 | leifmadsen | that's not a pattern.... |
| 13:44.40 | vizzy | extension 1 of my ivr menu |
| 13:44.45 | vizzy | and to store the key i use |
| 13:44.46 | vizzy | exten => _2,n,Read(NEWPIN,,4,1,5) |
| 13:44.46 | vizzy | exten => _2,n,Set(DB(/int_conference_passwd/room551)=${NEWPIN}) |
| 13:44.46 | vizzy | exten => _2,n,sendText("NEW PIN ${NEWPIN} FOR ROOM 501 SAVED") |
| 13:44.48 | leifmadsen | you don't need an underscore there |
| 13:44.58 | leifmadsen | and you have a priority 1 somewhere right? |
| 13:44.59 | DynamicFail | Has anyone heard of anything that allows asterisk configs to share everything so no trunking would be needed between multiple boxes running within your own lan |
| 13:45.03 | leifmadsen | pastebin the whole section |
| 13:45.20 | leifmadsen | DynamicFail: Asterisk SCF sounds like what you're looking for (not production ready) |
| 13:45.21 | *** join/#asterisk indika1978 (~chatzilla@203.115.2.202) |
| 13:45.24 | vizzy | sure i have prio one :) wait ill pastebin that |
| 13:45.35 | indika1978 | hi my festival service is not working |
| 13:46.05 | [TK]D-Fender | exten => _2,n,Set(DB(/int_conference_passwd/room551)=${NEWPIN}) <-- the DB shouldn't have a ? in front of the family... |
| 13:46.06 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 13:46.10 | [TK]D-Fender | <PROTECTED> |
| 13:46.46 | vizzy | ah! |
| 13:46.47 | vizzy | http://pastebin.com/87AdJUrP |
| 13:47.05 | indika1978 | it gave error ---http://pastebin.com/2SzCac1Y |
| 13:47.25 | indika1978 | pls help me |
| 13:48.53 | vizzy | so removing leading / should help? grml... again some strange error in one of the howtos/examples out there |
| 13:49.29 | [TK]D-Fender | exten => _2,n,Set(DB(/int_conference_passwd/room551)=${NEWPIN}) <- DB(family/key). No leading "/" |
| 13:49.30 | vizzy | i read through the asterisk book and it says the value of the key doesnt matter at all, the key is the passwd, not the value, .. thats wrong? |
| 13:49.34 | DynamicFail | leifmadsen, any idea how long before there is a release of asterisk SCF |
| 13:49.53 | [TK]D-Fender | vizzy, the complete value matters |
| 13:50.01 | vizzy | wow |
| 13:50.12 | vizzy | lemme test :D huge thx so far |
| 13:50.20 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 13:52.05 | leifmadsen | DynamicFail: no idea sorry |
| 13:52.50 | *** join/#asterisk Johnny- (~John@23-24-48-118-static.hfc.comcastbusiness.net) |
| 13:56.50 | vizzy | [TK]D-Fender, great, that works, i can save the key/value pair, BUT authentication wont work with that |
| 13:57.05 | vizzy | exten => _4,n,Authenticate(int_conference_passwd/room551,d,4) and i enter the correct key gives me a 'wrong passwd' |
| 13:57.43 | vizzy | thats what i mean, passwd is then 'room551' , if i got the docu right |
| 13:57.57 | vizzy | and not the value of key room551 |
| 13:58.07 | vizzy | am i correct here? |
| 13:58.16 | [TK]D-Fender | ziYou don't pass authenticate a KEYPAI |
| 13:58.19 | [TK]D-Fender | that is DB only |
| 13:58.25 | [TK]D-Fender | Yu pas it an actual VALUE |
| 13:58.36 | [TK]D-Fender | So go refernce the DB key to get the value there |
| 13:58.40 | vizzy | sorry i dont understand |
| 13:58.45 | vizzy | ah |
| 13:58.53 | [TK]D-Fender | Authenticate does NOT look at DB by itself <- |
| 13:59.13 | vizzy | ok, i have to substitute and get the key first, assign it to a var and use that var to authenticate? |
| 13:59.21 | vizzy | phew |
| 13:59.23 | [TK]D-Fender | You need to put an actual number there |
| 13:59.37 | vizzy | the number should be dynamic |
| 13:59.38 | [TK]D-Fender | No need to put it in a var. Yuo can pull it directly in-line there |
| 13:59.58 | vizzy | i dont know it, thats what i mean, the key should be editeable |
| 14:00.12 | vizzy | it should be stored in the db |
| 14:00.28 | vizzy | and the user who initiates the conference room should be able to change it |
| 14:00.48 | vizzy | so i cant put inline a value i dont know |
| 14:00.57 | vizzy | or have to change the script on every conference? |
| 14:01.29 | vizzy | ill use a variable so i dont have to mess with the weird authenticate syntax i guess |
| 14:02.18 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-oovpfcsjuohgannk) |
| 14:02.18 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:03.41 | vizzy | ok, auth wirks when assign the db value to a variable first :) |
| 14:04.03 | vizzy | but now i am asked for a pin of a conference room which has NO pin set, crazy |
| 14:06.47 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-tgxkuuzcbcqdqgys) |
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| 14:13.06 | [TK]D-Fender | vizzy, Maybe you should check first... |
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| 14:13.56 | [TK]D-Fender | <PROTECTED> |
| 14:21.14 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 14:21.32 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:21.56 | vizzy | ah, with DB, ... aaaah :) i got it |
| 14:22.09 | vizzy | i thought option d would do a lookup in the astdb itself |
| 14:22.28 | vizzy | hmmm, i have talk to my boss to get an asterisk book soon |
| 14:23.14 | vizzy | is there something available? i prefer paper, black on white .. instead of thousands of online docs i cant remember where i read a solution :) |
| 14:24.49 | vizzy | i guess the 'definitve guide to asterisk' is the one you guys wrote and will also earn something from it |
| 14:24.54 | vizzy | correct? |
| 14:25.27 | vizzy | but the cookbook has only 66 pages? |
| 14:26.25 | [TK]D-Fender | ~book |
| 14:26.25 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:26.43 | vizzy | yeah |
| 14:26.59 | vizzy | but what about the cookbook? does it contain only 66 pages? for 25 euro? |
| 14:27.01 | vizzy | wow |
| 14:30.28 | vizzy | amazon says the definitve guide has 695 pages, when i look at the single html online version i cant believe it will fill up 695 printed pages |
| 14:31.38 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:33.10 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 14:33.43 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 14:34.15 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:44.58 | leifmadsen | vizzy: yes, 695 pages |
| 14:45.10 | leifmadsen | vizzy: for cookbook I recommend non-print version as it's fairly light |
| 14:45.27 | *** join/#asterisk sklav (~sklav@unaffiliated/sklav) |
| 14:45.27 | leifmadsen | or just purchase ATDG and read cookbook online for free :) |
| 14:46.00 | leifmadsen | remember 695 pages includes appendices and index |
| 14:46.02 | sklav | Hi anybody here familiar with the dta310 from 8x8 or packet 8 |
| 14:46.18 | vizzy | ok, ill talk to my boss again, for now he says i should read online, ... wonder how he will look if i put hundreds of yellow postits around my monitor .... |
| 14:46.28 | leifmadsen | last chapter ends at page 595 |
| 14:47.10 | vizzy | leifmadsen, i prefer black on white printed on paper, can read everywhere, no need to be online all the time and switch from one chapter to another in browsertabs ;-) |
| 14:47.20 | leifmadsen | true story |
| 14:47.24 | *** part/#asterisk kaspar (~kaspar@p50811EF2.dip.t-dialin.net) |
| 14:47.49 | vizzy | ill get the book, maybe i have to pay it privately and then never give it away :D |
| 14:48.11 | bitglue | you know what they say...you can't burn a light emitting rectangle made of metal. |
| 14:49.36 | vizzy | oh, metal easily can burn..... :D |
| 14:50.11 | bitglue | you know what they say...you can't burn a metal rectangle full of lights without suplemental oxygen or increasing the surface area first. |
| 14:51.11 | drmessano | Is that anything like wrapping an inclined plane helically around an axis? |
| 14:52.38 | vizzy | i only use rockets |
| 14:56.48 | *** join/#asterisk F2Knight (~Adium@70-89-188-5-or.portland.hfc.comcastbusiness.net) |
| 14:57.53 | sklav | So anybody know or can reference me to a how-to that is recent that details how i may be able to unlock a packet 8 dta310 analog to digital converter |
| 14:58.17 | *** join/#asterisk pbxMan (c335d9a2@gateway/web/freenode/ip.195.53.217.162) |
| 14:58.40 | pbxMan | hello Asterisk people |
| 14:58.53 | pabelanger | hello Asterisk user |
| 14:59.00 | leifmadsen | :) |
| 15:00.33 | sklav | Hi guys is there a room that deals with hardware for asterisk? |
| 15:00.46 | leifmadsen | this one |
| 15:00.54 | sklav | i see |
| 15:01.08 | leifmadsen | I don't know how many people are going to be familiar with your packet 8 issue. That's very specific. |
| 15:01.09 | sklav | so nobody has any reference for my previous question |
| 15:01.15 | sklav | i see |
| 15:01.16 | leifmadsen | I'd be surprised if there were |
| 15:01.25 | F2Knight | Good morning asterisk lovers |
| 15:01.29 | leifmadsen | Google will be your best friend. If you can't find it there, I suspect that data doesnt exist. |
| 15:01.30 | F2Knight | morning leif |
| 15:01.36 | sklav | i have 3 devices i got 2 working with asterisk |
| 15:01.47 | sklav | problem is google has nothing for this specific issue |
| 15:02.32 | sklav | one has recent packet 8 firmware which i guess doesn't allow modifying the sip settings |
| 15:03.02 | sklav | no worries thank you anyways, just a shame to throw out perfectly good hardware |
| 15:05.43 | *** join/#asterisk uskerine (~abc@10.Red-2-138-179.dynamicIP.rima-tde.net) |
| 15:05.45 | uskerine | hi |
| 15:06.21 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 15:06.24 | uskerine | i have installed a ISDN PRI in an asterisk server, so far I managed to configure everything in a test server using analog lines. I have configured dahdi_conf as indicated in the doc of the digium card and led is green |
| 15:06.25 | [TK]D-Fender | sklav, If yuo have to throw it away ... perhaps it isn't "perfectly good", just "marginally good for one restricted thing" |
| 15:06.27 | uskerine | how can i start debugging? |
| 15:06.40 | vizzy | thx and goodbye |
| 15:06.56 | [TK]D-Fender | uskerine, What tells yuo there are bugs? |
| 15:07.22 | [TK]D-Fender | uskerine, So far you have not spoken of having any actual problem. |
| 15:07.23 | sruffell | uskerine: I recommend try developing using before jumping straight into debugging. Isn't that a little pessimistic? ;) |
| 15:07.27 | uskerine | i call on of the assigned numbers to the E1 and I got the message "Telefonica informs you that there is overload in the private number assigned." |
| 15:07.33 | sklav | [TK]D-Fender the device works but i prefer to use it with asterisk now and the sip stuff is hardcoded to point to 8x8 |
| 15:07.44 | uskerine | basically I don't know where to start |
| 15:07.52 | [TK]D-Fender | uskerine, "pri debug span X" <- enable for your span. Look at what comes in. |
| 15:07.57 | uskerine | ok thanks! |
| 15:08.22 | [TK]D-Fender | sklav, They locked ti specifically to keep you out. Don't be too disappointed if they succeeded in their goal. |
| 15:08.31 | sklav | :) |
| 15:08.42 | [TK]D-Fender | or "surprised" at least |
| 15:08.48 | sklav | i try not to give up to easily |
| 15:09.02 | sklav | 2 out of 3 aint bad i guess |
| 15:09.09 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 15:09.10 | [TK]D-Fender | </meatloaf> |
| 15:09.28 | bitglue | sklav: perhaps you can play tricks on it with your network hardware to redirect its traffic to 8x8 to the destination of your choosing. |
| 15:10.03 | bitglue | sklav: like, NAT. |
| 15:10.14 | uskerine | pbxalfa*CLI> pri show spans |
| 15:10.14 | uskerine | No such command 'pri show spans' |
| 15:10.15 | F2Knight | some IPTABLES magic… but the destination should probably be listening to the same name/ip |
| 15:10.20 | sklav | agreed but i cannot even set the userid |
| 15:10.44 | F2Knight | uskerine: try dahdi |
| 15:11.10 | sklav | i would have to check wireshark to see what it does or sends as info and take it from there |
| 15:11.19 | sklav | brb |
| 15:12.14 | uskerine | F2Knight, I don't have dahdi module loaded |
| 15:12.18 | uskerine | (bad start) |
| 15:12.35 | F2Knight | uskerine: w/o your dahdi I do not believe the PRI will load. |
| 15:12.49 | F2Knight | try 'module load chan_dahdi' to load dahdi |
| 15:12.59 | uskerine | what enables chan_dahdi module to be loaded by default? |
| 15:13.16 | F2Knight | if it wont load you may need to check your dahdi config and the stuff before asterisk gets it |
| 15:13.35 | uskerine | i can't find in while TABbing module load chan_ |
| 15:13.40 | sruffell | without dahdi loaded, I wonder how the message from Telefonica was even making it to the server. |
| 15:13.42 | F2Knight | in modules.conf if autoload = yes then it should load but it will fail to load if it is not properly configured |
| 15:14.16 | F2Knight | if you manually load modules as I do , then you need to add a load => chan_dadhdi.so |
| 15:14.24 | uskerine | sruffell the message from telefonica was voice message delivered by telefonica when i was calling from an external phone |
| 15:14.47 | sruffell | ahh..ok, that makes sense then…can't route the call because the line to your server isn't up. |
| 15:14.59 | F2Knight | okay .. uskerine I got a few moments to point you down the right path |
| 15:15.04 | F2Knight | was the PRI working before? |
| 15:15.05 | sruffell | ..which F2Knight appears to be helping you with.. |
| 15:15.17 | uskerine | no F2Knight, it is my first PRI installed on a fresh new server |
| 15:15.20 | F2Knight | okay |
| 15:15.37 | F2Knight | ASSuming you have done nothing but plug it in to the computer and turned it on. |
| 15:15.50 | uskerine | i can't see any chan_dahdi in modules.conf |
| 15:15.50 | F2Knight | you need to generate a config file for the dahdi drivers. |
| 15:15.57 | uskerine | well, dahdi_tool shows card up and sync |
| 15:16.00 | F2Knight | exit asterisk |
| 15:16.13 | uskerine | and i configured dahdi_conf as indicated by digium card manual |
| 15:16.14 | uskerine | ok |
| 15:16.17 | uskerine | i exit asterisk |
| 15:16.23 | F2Knight | and at the linux commnad prompt |
| 15:16.39 | F2Knight | my suggestion would be to back up the /etc/dahdi directory |
| 15:16.46 | uskerine | ok |
| 15:16.47 | uskerine | i will do that |
| 15:16.59 | uskerine | done |
| 15:17.04 | uskerine | i actually modified system.conf |
| 15:17.12 | uskerine | -i did a backup of the file- |
| 15:17.15 | F2Knight | and use the dahdi_genconf command |
| 15:17.18 | uskerine | ok |
| 15:17.29 | F2Knight | it will overwrite your settings.. /etc/dahdi/system.conf |
| 15:17.44 | uskerine | ok |
| 15:17.48 | F2Knight | what that does is some autodetect magic to see what is IN the system. |
| 15:17.49 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 15:17.51 | uskerine | done |
| 15:18.08 | F2Knight | do a lsmod |
| 15:18.17 | F2Knight | and look to make sure dahdi is loaded |
| 15:18.21 | F2Knight | it may or may not be |
| 15:18.23 | uskerine | http://pastebin.com/djavy0Qn |
| 15:18.31 | uskerine | system.conf generated |
| 15:18.45 | uskerine | does it looks fine for an ISDN PRI? |
| 15:19.32 | uskerine | lsmod | grep dah -> shows dahdi_transcode, dahdi_dummy, dahdi_voicebus, dahdi and crc_ccitt |
| 15:19.51 | F2Knight | that looks good.. |
| 15:19.56 | F2Knight | did you edit the system.conf? |
| 15:20.09 | uskerine | yes i did, following the manual of the digium card |
| 15:20.17 | uskerine | but what i showed you was the new generated system.conf |
| 15:20.30 | uskerine | using dahdi_genconf |
| 15:20.33 | F2Knight | you have installed the oslec echo canceler? |
| 15:20.47 | F2Knight | umm… never seen the tool detect that automatically before... |
| 15:20.51 | uskerine | i have hardware chancellation child board |
| 15:21.17 | uskerine | by "have you installed" you mean if I put something in the system.conf? |
| 15:21.29 | uskerine | or if i have physically the board? |
| 15:21.47 | F2Knight | by install i mean you downloaded and installed the package.. that i do not believe is packaged with Asterisk. |
| 15:22.17 | uskerine | i don't think so |
| 15:22.24 | F2Knight | 31 channels on your PRI? does that sound right? |
| 15:22.27 | uskerine | i installed binary asterisk distribution in ubuntu server 12.04 |
| 15:22.48 | uskerine | a PRI is 32 channels, one signalling and one sync, right? |
| 15:23.06 | F2Knight | a pri is 24 channels in the US |
| 15:23.10 | uskerine | i am in europe |
| 15:23.12 | uskerine | it is an E1 |
| 15:23.15 | uskerine | <PROTECTED> |
| 15:23.23 | uskerine | 0 for sync, 16 for signalling |
| 15:23.31 | F2Knight | the dchan is usually on the end not the middle. |
| 15:23.47 | F2Knight | okay so E1 are a little different. |
| 15:23.51 | pabelanger | yes |
| 15:23.56 | uskerine | as far as I know E1 is channel 16 |
| 15:24.13 | F2Knight | but same process.. just verify that the information the sync and all is correct with your provider. (they should have given you that information.) |
| 15:24.38 | F2Knight | if your not in the US then 1.# Global data |
| 15:24.38 | F2Knight | 2. |
| 15:24.38 | F2Knight | 3.loadzone = us |
| 15:24.38 | F2Knight | 4.defaultzone = us |
| 15:24.38 | F2Knight | should be changed |
| 15:25.24 | uskerine | span=1,1,0,ccs,hdb3,crc4 |
| 15:25.27 | [TK]D-Fender | Sync isn't a channel. |
| 15:25.30 | [TK]D-Fender | ~e1 |
| 15:25.30 | infobot | [~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong (who use T1 and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling. |
| 15:25.32 | [TK]D-Fender | ^ |
| 15:25.44 | [TK]D-Fender | ~pri |
| 15:25.44 | infobot | pri is probably [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 15:25.58 | uskerine | where can i check here that card is taking reference from the ISDN PRI ? |
| 15:26.14 | uskerine | provider details are: |
| 15:26.25 | uskerine | - PBX must be clocked from our circuit |
| 15:26.31 | uskerine | - EuroISDN-DSS1 protocol |
| 15:26.36 | [TK]D-Fender | <uskerine> span=1,1,0,ccs,hdb3,crc4 <--- other side = primary timing source |
| 15:26.37 | uskerine | - E.164 numbering plan |
| 15:26.48 | [TK]D-Fender | uskerine, Where is the debug>? |
| 15:26.52 | uskerine | ok |
| 15:26.55 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 15:26.57 | uskerine | i will try the debug |
| 15:27.22 | F2Knight | [TK]D-Fender: he doesn't have the dahdi drivers loaded in to asterisk.. |
| 15:27.36 | sruffell | uskerine: When you loaded the driver you will want to use (with 2.6.1) "modprobe wcte12xp default_linemode=e1", then when you run dahdi_genconf the parameters will be correct. |
| 15:28.13 | sruffell | without setting the jumpers for E1 on the card, or using the default_linemode module parameter…the card won't work on an E1 line. |
| 15:28.16 | uskerine | i don't have dahdi drivers loaded to asterisk |
| 15:28.19 | uskerine | that's right |
| 15:28.23 | *** part/#asterisk sklav (~sklav@unaffiliated/sklav) |
| 15:28.57 | F2Knight | sruffell: may be very right about that , as I do not use E1 cards they are a little different. make sure you add the module wct…. to the /etc/dahdi/modules.conf |
| 15:29.07 | sruffell | doh..never mind…I thought I saw your pastebin was configured for T1…never mind me and sorry for stirring the pot. |
| 15:29.25 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 15:29.29 | [TK]D-Fender | uskerine, So go fix that |
| 15:29.34 | uskerine | i am fixing it |
| 15:29.37 | F2Knight | it was thats why I was thinking he didn't run the command correctly .. of for some reason the card thinks its in the good old USA |
| 15:29.42 | uskerine | in my temp server i have wctdm24xxp |
| 15:29.48 | uskerine | and that's missing in the new server |
| 15:29.52 | uskerine | is it same driver? |
| 15:29.55 | uskerine | it is different card |
| 15:30.14 | uskerine | no |
| 15:30.17 | uskerine | it shoud be wcte12xp |
| 15:30.26 | [TK]D-Fender | uskerine, Your channel confis are E1 standard. |
| 15:30.38 | [TK]D-Fender | Go fix the * side. With No DAHDI you aren't getting anywhere. |
| 15:30.42 | [TK]D-Fender | Fix that then continue with us |
| 15:31.04 | F2Knight | uskerine: what does dahdi_toolshow you ? |
| 15:31.15 | uskerine | do i have to restart the server? |
| 15:31.28 | uskerine | <PROTECTED> |
| 15:31.31 | uskerine | but nothing happens |
| 15:31.46 | F2Knight | that is because asterisk does not load the dirver |
| 15:31.50 | F2Knight | the OS loads the driver |
| 15:31.56 | F2Knight | asterisk looks to see if the driver is loaded |
| 15:32.02 | uskerine | F2Knight regarding E1/T1 selection, i changed the jumper to E1 |
| 15:32.07 | F2Knight | do a modprob and the driver name to load it by hand |
| 15:32.11 | uskerine | iok |
| 15:32.24 | F2Knight | you changed the jumper on a hot system? |
| 15:32.57 | uskerine | no, i changed it when i installed the server, some weeks ago |
| 15:32.57 | uskerine | :) |
| 15:33.04 | F2Knight | okay. |
| 15:33.08 | uskerine | i will restart the server, modprobe does not work |
| 15:33.09 | uskerine | hold on |
| 15:33.16 | uskerine | is that fine? |
| 15:33.22 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-vqlufdrtppulxrdq) |
| 15:33.22 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:33.26 | F2Knight | if no ones using the system sure.. |
| 15:34.13 | uskerine | rebooting |
| 15:34.15 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 15:35.45 | *** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 15:36.10 | uskerine | remote eyes and hands restarting the server :) |
| 15:38.06 | uskerine | no way |
| 15:38.10 | uskerine | dahdi is still not there |
| 15:38.43 | uskerine | sysadmin@pbxalfa:/etc/dahdi$ cat modules shows --->> dahdi, dahdi_dummy, dahdi_transcode and recently added wcte12xp |
| 15:39.00 | uskerine | mm |
| 15:39.05 | uskerine | lsmod shows no wcte12xp or dahdi |
| 15:41.42 | uskerine | sudo modprobe wcte12xp |
| 15:41.55 | uskerine | WARNING: Error inserting dahdi_voicebus (/lib/modules/3.2.0-29-generic/updates/dkms/dahdi_voicebus.ko): Invalid module format |
| 15:42.02 | uskerine | FATAL: Error inserting wcte12xp (/lib/modules/3.2.0-29-generic/updates/dkms/wcte12xp.ko): Invalid module format |
| 15:42.09 | uskerine | it was loaded before rebooting |
| 15:42.34 | uskerine | any idea? |
| 15:43.18 | sruffell | <— doesn't use the dkms drivers. |
| 15:43.28 | sruffell | I always just uninstall them and build from source. |
| 15:43.28 | pabelanger | sruffell: you are missing out |
| 15:43.35 | sruffell | I think not. |
| 15:43.37 | sruffell | :) |
| 15:44.02 | pabelanger | uskerine: what OS? |
| 15:44.07 | uskerine | ubuntu server 12.04 |
| 15:44.16 | uskerine | binary installation of asterisk |
| 15:44.28 | pabelanger | sudo dpkg-reconfigure dahdi |
| 15:44.31 | uskerine | ok |
| 15:44.32 | pabelanger | should rebuild them |
| 15:44.34 | pabelanger | I think |
| 15:44.52 | uskerine | i also did sudo apt-get upgrade |
| 15:44.55 | uskerine | (just in case) |
| 15:45.01 | pabelanger | have you rebooted since? |
| 15:45.16 | uskerine | i will |
| 15:45.30 | pabelanger | that should force the modules to rebuild if out of since |
| 15:45.34 | pabelanger | sync* |
| 15:46.22 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.128.111) |
| 15:47.13 | F2Knight | uskerine: when you say binary install are you installing from added digium repos or the Ubuntu ones? |
| 15:47.42 | F2Knight | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 15:47.46 | uskerine | FATAL: Error inserting dahdi (/lib/modules/3.2.0-29-generic/updates/dkms/dahdi.ko): Invalid module format |
| 15:47.48 | uskerine | same stuff |
| 15:47.53 | uskerine | ubuntu iones |
| 15:48.02 | F2Knight | I believe Ubuntu has a lot older version and digium is at 1.8 |
| 15:48.10 | F2Knight | 1.8.11 on the repo as I recall |
| 15:48.33 | F2Knight | that error sounds like you do not have the correct build. |
| 15:49.00 | F2Knight | is this a new build? |
| 15:50.02 | uskerine | i am a bit lost |
| 15:50.06 | uskerine | i did apt-get upgrade |
| 15:50.08 | uskerine | then this happens |
| 15:50.24 | uskerine | i have followed purge/install as indicated in the link you have shared |
| 15:51.02 | mazpe | any have has a particular favor package firewall that plays well with asterisk or that they prefered? like (pfsense, m0n0wall, ipcop?) |
| 15:51.50 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:53.16 | uskerine | W: Failed to fetch http://packages.asterisk.org/deb/dists/precise/main/source/Sources 404 Not Found |
| 15:56.00 | jpsharp | packages suck :) |
| 15:57.58 | uskerine | :( |
| 16:00.36 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-oovpfcsjuohgannk) |
| 16:00.47 | uskerine | ok |
| 16:00.48 | uskerine | no way |
| 16:00.53 | uskerine | any idea on what should i do? |
| 16:00.57 | uskerine | i insalled asterisk-1.8 |
| 16:01.15 | uskerine | no way to add asterisk repositories as they seem to be only for oneiric, not for precise (12.04) |
| 16:02.51 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-193.mobile.uci.edu) |
| 16:04.11 | pabelanger | Right |
| 16:05.19 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-xjfohznpambcglan) |
| 16:05.29 | pabelanger | uskerine: I roll my own packages for 12.04, if you want to test using them. See if they work |
| 16:05.33 | pabelanger | https://github.com/kickstandproject/asterisk/wiki/Installing-Packages |
| 16:05.44 | pabelanger | jpsharp: how many asterisk boxes do you manage? |
| 16:06.31 | uskerine | this one |
| 16:06.53 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 16:16.03 | *** join/#asterisk jakent (~jakent@c-71-63-6-140.hsd1.va.comcast.net) |
| 16:21.02 | uskerine | pabelanger, i followed the install from your repository, but it keeps failing |
| 16:21.08 | uskerine | FATAL : Error inserting wcte12xp |
| 16:21.15 | uskerine | (modprobe wcte12xp) |
| 16:21.48 | sruffell | same error about the module format? |
| 16:22.09 | uskerine | invalid module format |
| 16:22.11 | uskerine | yeah |
| 16:22.27 | uskerine | i am purging asterisk* and dahdi* and reinstalling it again to be sure there is no conflict |
| 16:22.31 | sruffell | yeah..whenever I've seen these problems it's because there are now two copies of the driver in /lib/modules/`uname -r` |
| 16:22.56 | sruffell | What is the output of find /lib/modules/`uname -r` -name "wcte12xp.ko" |
| 16:23.02 | uskerine | let me check |
| 16:23.20 | uskerine | now it looks better |
| 16:23.21 | uskerine | like |
| 16:23.26 | uskerine | "First installation: checking all kernels...." |
| 16:23.31 | uskerine | building only for 3.2.0-29-generic |
| 16:23.35 | uskerine | building inital module.... |
| 16:23.39 | uskerine | smells good :) |
| 16:23.54 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 16:23.59 | uskerine | (I purged, added repository and reinstalled asterisk and dahdi) |
| 16:24.18 | sruffell | waits until you say you can load the driver into your kernel successfully. |
| 16:24.46 | uskerine | sysadmin@pbxalfa:~$ lsmod | grep wcte12xp |
| 16:24.47 | uskerine | wcte12xp 36910 0 |
| 16:24.53 | sruffell | progress! |
| 16:24.59 | uskerine | that little bastard is loaded now |
| 16:25.17 | uskerine | but dahdi is still not present in asterisk |
| 16:25.24 | uskerine | oh my god, this is endless |
| 16:25.34 | sruffell | and $ dahdi_scan | grep type" says digital-E1? |
| 16:26.01 | uskerine | i added it to /etc/dahdi/modules again |
| 16:26.05 | uskerine | i will check, i am rebooting now |
| 16:26.12 | uskerine | hold on |
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| 16:28.11 | uskerine | devicetype=Wildcard TE121 |
| 16:28.11 | uskerine | type=digital-E1 |
| 16:28.38 | uskerine | but i think i don't have dahdi in asterisk |
| 16:28.46 | uskerine | i see wcte12xp now in lsmod |
| 16:29.06 | uskerine | no chan_dahdi, no commands starting with dahdi |
| 16:29.23 | uskerine | maybe i have to install asterisk-dahdi |
| 16:29.25 | uskerine | ? |
| 16:30.27 | sruffell | someone who knows about the packages will probably know better than me. I would just make sure dahdi is configured (did you run dahdi_cfg?) that the config files are present in /etc/asterisk and if so, from the asterisk cli, run "module load chan_dahdi.so" and see what the error is. |
| 16:31.33 | uskerine | i just ran dahdi_cfg |
| 16:31.51 | sruffell | BUT…before even starting asterisk..make sure your card is not in any alarm state. |
| 16:31.57 | uskerine | the problem is that |
| 16:32.04 | uskerine | there is no chan_dahdi.so |
| 16:32.06 | uskerine | in asterisk |
| 16:32.14 | sruffell | ok…then, there must be some other package piece that is missing. |
| 16:33.10 | bitglue | so, i've read in several places that I should get something neat at http://localhost:8088/asterisk/static/ajamdemo.html, but all I can get out of the HTTP server is a 404 response. How do I go about troubleshooting that? |
| 16:33.15 | sruffell | …and I always just build from source since I don't know the details of package management on debian too well. |
| 16:33.37 | uskerine | ok |
| 16:33.44 | uskerine | is difficult to build from soyrce |
| 16:33.44 | uskerine | = |
| 16:33.46 | uskerine | ? |
| 16:34.15 | bitglue | in fact, i can't get any URL besides /asterisk/httpstatus to display anything besides an error. I'm kinda wondering how I verify anything is working. |
| 16:34.28 | uskerine | good |
| 16:34.39 | uskerine | i did sudo apt-get install asterisk-dahdi |
| 16:34.45 | uskerine | now chan_dahdi.so is present |
| 16:35.01 | uskerine | but dahdi_config shows board as "UNCONFIGURED" |
| 16:47.34 | [TK]D-Fender | So go configure it |
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| 16:51.05 | bitglue | so eh, are none of the static http files distributed with the digium asterisk packages? It's looking that way to me...I have a /usr/share/asterisk/static-http but it is empty. |
| 16:51.59 | leifmadsen | bitglue: no because that would probably be populated via asterisk-gui |
| 16:52.02 | leifmadsen | not asterisk proper |
| 16:53.17 | bitglue | leifmadsen: i'm mostly interesting in this "ajamdemo.html" thing i'm reading about. Any idea where I get it? |
| 16:53.30 | leifmadsen | bitglue: asterisk-gui |
| 16:53.39 | bitglue | leifmadsen: is that available under a name that's searchable? |
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| 16:54.09 | leifmadsen | bitglue: ... |
| 16:54.31 | leifmadsen | did you try looking at the available RPMs? http://packages.asterisk.org/centos/5/current/i386/RPMS/ |
| 16:54.35 | bitglue | leifmadsen: well, searching for "asterisk-gui" turns up about 100 asterisk gui projects |
| 16:54.38 | leifmadsen | because 'asterisk-gui' is about as searchable as it gets |
| 16:54.56 | bitglue | didn't look there; running debian :( |
| 16:55.10 | bitglue | there is no asterisk-gui package at least for debian |
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| 16:55.24 | leifmadsen | bitglue: then go up a couple directories |
| 16:55.36 | leifmadsen | but the debian packages are quite old now it looks like |
| 16:55.44 | leifmadsen | could just install from subversion too |
| 16:55.54 | leifmadsen | beyond that, you're on your own as I don't use any GUIs |
| 16:56.34 | bitglue | ok, but what subversion? |
| 16:56.41 | leifmadsen | asterisk-gui |
| 16:56.47 | bitglue | ok, *where*? |
| 16:56.58 | bitglue | try searching google for asterisk-gui and you will see how impossible it is to find any information on it |
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| 16:57.39 | bitglue | is there like...a project page? |
| 16:57.43 | leifmadsen | asterisk.org? |
| 16:57.46 | bitglue | is it part of asterisk? |
| 16:58.17 | leifmadsen | no |
| 16:58.20 | leifmadsen | svn.asterisk.org/svn |
| 16:59.07 | leifmadsen | wiki.asterisk.org and then I typed "gui" and data on it was the first link |
| 17:00.14 | bitglue | well, that's good to know |
| 17:00.20 | bitglue | it's still damn hard to find |
| 17:00.41 | bitglue | it's like trying to find information on a particular model of electric car named "electric car" when you don't know who makes it or where you get one. |
| 17:00.49 | leifmadsen | I guess.... |
| 17:01.05 | bitglue | http://xkcd.com/993/ |
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